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 example: mailboxes=6001@default.
275 For endpoints that SUBSCRIBE for MWI, you can set the <literal>mailboxes</literal> option in your AOR
277 </para></description>
279 <configOption name="moh_suggest" default="default">
280 <synopsis>Default Music On Hold class</synopsis>
282 <configOption name="outbound_auth">
283 <synopsis>Authentication object used for outbound requests</synopsis>
285 <configOption name="outbound_proxy">
286 <synopsis>Proxy through which to send requests, a full SIP URI must be provided</synopsis>
288 <configOption name="rewrite_contact">
289 <synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
291 On inbound SIP messages from this endpoint, the Contact header will be changed to have the
292 source IP address and port. This option does not affect outbound messages send to this
294 </para></description>
296 <configOption name="rtp_ipv6" default="no">
297 <synopsis>Allow use of IPv6 for RTP traffic</synopsis>
299 <configOption name="rtp_symmetric" default="no">
300 <synopsis>Enforce that RTP must be symmetric</synopsis>
302 <configOption name="send_diversion" default="yes">
303 <synopsis>Send the Diversion header, conveying the diversion
304 information to the called user agent</synopsis>
306 <configOption name="send_pai" default="no">
307 <synopsis>Send the P-Asserted-Identity header</synopsis>
309 <configOption name="send_rpid" default="no">
310 <synopsis>Send the Remote-Party-ID header</synopsis>
312 <configOption name="timers_min_se" default="90">
313 <synopsis>Minimum session timers expiration period</synopsis>
315 Minimium session timer expiration period. Time in seconds.
316 </para></description>
318 <configOption name="timers" default="yes">
319 <synopsis>Session timers for SIP packets</synopsis>
322 <enum name="forced" />
324 <enum name="required" />
329 <configOption name="timers_sess_expires" default="1800">
330 <synopsis>Maximum session timer expiration period</synopsis>
332 Maximium session timer expiration period. Time in seconds.
333 </para></description>
335 <configOption name="transport">
336 <synopsis>Desired transport configuration</synopsis>
338 This will set the desired transport configuration to send SIP data through.
340 <warning><para>Not specifying a transport will <emphasis>DEFAULT</emphasis>
341 to the first configured transport in <filename>pjsip.conf</filename> which is
342 valid for the URI we are trying to contact.
344 <warning><para>Transport configuration is not affected by reloads. In order to
345 change transports, a full Asterisk restart is required</para></warning>
348 <configOption name="trust_id_inbound" default="no">
349 <synopsis>Accept identification information received from this endpoint</synopsis>
350 <description><para>This option determines whether Asterisk will accept
351 identification from the endpoint from headers such as P-Asserted-Identity
352 or Remote-Party-ID header. This option applies both to calls originating from the
353 endpoint and calls originating from Asterisk. If <literal>no</literal>, the
354 configured Caller-ID from pjsip.conf will always be used as the identity for
355 the endpoint.</para></description>
357 <configOption name="trust_id_outbound" default="no">
358 <synopsis>Send private identification details to the endpoint.</synopsis>
359 <description><para>This option determines whether res_pjsip will send private
360 identification information to the endpoint. If <literal>no</literal>,
361 private Caller-ID information will not be forwarded to the endpoint.
362 "Private" in this case refers to any method of restricting identification.
363 Example: setting <replaceable>callerid_privacy</replaceable> to any
364 <literal>prohib</literal> variation.
365 Example: If <replaceable>trust_id_inbound</replaceable> is set to
366 <literal>yes</literal>, the presence of a <literal>Privacy: id</literal>
367 header in a SIP request or response would indicate the identification
368 provided in the request is private.</para></description>
370 <configOption name="type">
371 <synopsis>Must be of type 'endpoint'.</synopsis>
373 <configOption name="use_ptime" default="no">
374 <synopsis>Use Endpoint's requested packetisation interval</synopsis>
376 <configOption name="use_avpf" default="no">
377 <synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this
380 If set to <literal>yes</literal>, res_pjsip will use use the AVPF or SAVPF RTP
381 profile for all media offers on outbound calls and media updates and will
382 decline media offers not using the AVPF or SAVPF profile.
384 If set to <literal>no</literal>, res_pjsip will use use the AVP or SAVP RTP
385 profile for all media offers on outbound calls and media updates and will
386 decline media offers not using the AVP or SAVP profile.
387 </para></description>
389 <configOption name="media_encryption" default="no">
390 <synopsis>Determines whether res_pjsip will use and enforce usage of media encryption
391 for this endpoint.</synopsis>
394 <enum name="no"><para>
395 res_pjsip will offer no encryption and allow no encryption to be setup.
397 <enum name="sdes"><para>
398 res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP
399 transport should be used in conjunction with this option to prevent
400 exposure of media encryption keys.
402 <enum name="dtls"><para>
403 res_pjsip will offer DTLS-SRTP setup.
408 <configOption name="inband_progress" default="no">
409 <synopsis>Determines whether chan_pjsip will indicate ringing using inband
412 If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
413 when told to indicate ringing and will immediately start sending ringing
416 If set to <literal>no</literal>, chan_pjsip will send a 180 Ringing when told
417 to indicate ringing and will NOT send it as audio.
418 </para></description>
420 <configOption name="call_group">
421 <synopsis>The numeric pickup groups for a channel.</synopsis>
423 Can be set to a comma separated list of numbers or ranges between the values
424 of 0-63 (maximum of 64 groups).
425 </para></description>
427 <configOption name="pickup_group">
428 <synopsis>The numeric pickup groups that a channel can pickup.</synopsis>
430 Can be set to a comma separated list of numbers or ranges between the values
431 of 0-63 (maximum of 64 groups).
432 </para></description>
434 <configOption name="named_call_group">
435 <synopsis>The named pickup groups for a channel.</synopsis>
437 Can be set to a comma separated list of case sensitive strings limited by
438 supported line length.
439 </para></description>
441 <configOption name="named_pickup_group">
442 <synopsis>The named pickup groups that a channel can pickup.</synopsis>
444 Can be set to a comma separated list of case sensitive strings limited by
445 supported line length.
446 </para></description>
448 <configOption name="device_state_busy_at" default="0">
449 <synopsis>The number of in-use channels which will cause busy to be returned as device state</synopsis>
451 When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the
452 PJSIP channel driver will return busy as the device state instead of in use.
453 </para></description>
455 <configOption name="t38_udptl" default="no">
456 <synopsis>Whether T.38 UDPTL support is enabled or not</synopsis>
458 If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted
460 </para></description>
462 <configOption name="t38_udptl_ec" default="none">
463 <synopsis>T.38 UDPTL error correction method</synopsis>
466 <enum name="none"><para>
467 No error correction should be used.
469 <enum name="fec"><para>
470 Forward error correction should be used.
472 <enum name="redundancy"><para>
473 Redundacy error correction should be used.
478 <configOption name="t38_udptl_maxdatagram" default="0">
479 <synopsis>T.38 UDPTL maximum datagram size</synopsis>
481 This option can be set to override the maximum datagram of a remote endpoint for broken
483 </para></description>
485 <configOption name="fax_detect" default="no">
486 <synopsis>Whether CNG tone detection is enabled</synopsis>
488 This option can be set to send the session to the fax extension when a CNG tone is
490 </para></description>
492 <configOption name="t38_udptl_nat" default="no">
493 <synopsis>Whether NAT support is enabled on UDPTL sessions</synopsis>
495 When enabled the UDPTL stack will send UDPTL packets to the source address of
497 </para></description>
499 <configOption name="t38_udptl_ipv6" default="no">
500 <synopsis>Whether IPv6 is used for UDPTL Sessions</synopsis>
502 When enabled the UDPTL stack will use IPv6.
503 </para></description>
505 <configOption name="tone_zone">
506 <synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis>
508 <configOption name="language">
509 <synopsis>Set the default language to use for channels created for this endpoint.</synopsis>
511 <configOption name="one_touch_recording" default="no">
512 <synopsis>Determines whether one-touch recording is allowed for this endpoint.</synopsis>
514 <ref type="configOption">recordonfeature</ref>
515 <ref type="configOption">recordofffeature</ref>
518 <configOption name="record_on_feature" default="automixmon">
519 <synopsis>The feature to enact when one-touch recording is turned on.</synopsis>
521 <para>When an INFO request for one-touch recording arrives with a Record header set to "on", this
522 feature will be enabled for the channel. The feature designated here can be any built-in
523 or dynamic feature defined in features.conf.</para>
524 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
527 <ref type="configOption">one_touch_recording</ref>
528 <ref type="configOption">recordofffeature</ref>
531 <configOption name="record_off_feature" default="automixmon">
532 <synopsis>The feature to enact when one-touch recording is turned off.</synopsis>
534 <para>When an INFO request for one-touch recording arrives with a Record header set to "off", this
535 feature will be enabled for the channel. The feature designated here can be any built-in
536 or dynamic feature defined in features.conf.</para>
537 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
540 <ref type="configOption">one_touch_recording</ref>
541 <ref type="configOption">recordonfeature</ref>
544 <configOption name="rtp_engine" default="asterisk">
545 <synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis>
547 <configOption name="allow_transfer" default="yes">
548 <synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis>
550 <configOption name="sdp_owner" default="-">
551 <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
553 <configOption name="sdp_session" default="Asterisk">
554 <synopsis>String used for the SDP session (s=) line.</synopsis>
556 <configOption name="tos_audio">
557 <synopsis>DSCP TOS bits for audio streams</synopsis>
559 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
560 </para></description>
562 <configOption name="tos_video">
563 <synopsis>DSCP TOS bits for video streams</synopsis>
565 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
566 </para></description>
568 <configOption name="cos_audio">
569 <synopsis>Priority for audio streams</synopsis>
571 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
572 </para></description>
574 <configOption name="cos_video">
575 <synopsis>Priority for video streams</synopsis>
577 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
578 </para></description>
580 <configOption name="allow_subscribe" default="yes">
581 <synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis>
583 <configOption name="sub_min_expiry" default="60">
584 <synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis>
586 <configOption name="from_user">
587 <synopsis>Username to use in From header for requests to this endpoint.</synopsis>
589 <configOption name="mwi_from_user">
590 <synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis>
592 <configOption name="from_domain">
593 <synopsis>Domain to user in From header for requests to this endpoint.</synopsis>
595 <configOption name="dtls_verify">
596 <synopsis>Verify that the provided peer certificate is valid</synopsis>
598 This option only applies if <replaceable>media_encryption</replaceable> is
599 set to <literal>dtls</literal>.
600 </para></description>
602 <configOption name="dtls_rekey">
603 <synopsis>Interval at which to renegotiate the TLS session and rekey the SRTP session</synopsis>
605 This option only applies if <replaceable>media_encryption</replaceable> is
606 set to <literal>dtls</literal>.
608 If this is not set or the value provided is 0 rekeying will be disabled.
609 </para></description>
611 <configOption name="dtls_cert_file">
612 <synopsis>Path to certificate file to present to peer</synopsis>
614 This option only applies if <replaceable>media_encryption</replaceable> is
615 set to <literal>dtls</literal>.
616 </para></description>
618 <configOption name="dtls_private_key">
619 <synopsis>Path to private key for certificate file</synopsis>
621 This option only applies if <replaceable>media_encryption</replaceable> is
622 set to <literal>dtls</literal>.
623 </para></description>
625 <configOption name="dtls_cipher">
626 <synopsis>Cipher to use for DTLS negotiation</synopsis>
628 This option only applies if <replaceable>media_encryption</replaceable> is
629 set to <literal>dtls</literal>.
631 Many options for acceptable ciphers. See link for more:
632 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
633 </para></description>
635 <configOption name="dtls_ca_file">
636 <synopsis>Path to certificate authority certificate</synopsis>
638 This option only applies if <replaceable>media_encryption</replaceable> is
639 set to <literal>dtls</literal>.
640 </para></description>
642 <configOption name="dtls_ca_path">
643 <synopsis>Path to a directory containing certificate authority certificates</synopsis>
645 This option only applies if <replaceable>media_encryption</replaceable> is
646 set to <literal>dtls</literal>.
647 </para></description>
649 <configOption name="dtls_setup">
650 <synopsis>Whether we are willing to accept connections, connect to the other party, or both.</synopsis>
653 This option only applies if <replaceable>media_encryption</replaceable> is
654 set to <literal>dtls</literal>.
657 <enum name="active"><para>
658 res_pjsip will make a connection to the peer.
660 <enum name="passive"><para>
661 res_pjsip will accept connections from the peer.
663 <enum name="actpass"><para>
664 res_pjsip will offer and accept connections from the peer.
669 <configOption name="srtp_tag_32">
670 <synopsis>Determines whether 32 byte tags should be used instead of 80 byte tags.</synopsis>
672 This option only applies if <replaceable>media_encryption</replaceable> is
673 set to <literal>sdes</literal> or <literal>dtls</literal>.
674 </para></description>
676 <configOption name="set_var">
677 <synopsis>Variable set on a channel involving the endpoint.</synopsis>
679 When a new channel is created using the endpoint set the specified
680 variable(s) on that channel. For multiple channel variables specify
681 multiple 'set_var'(s).
682 </para></description>
685 <configObject name="auth">
686 <synopsis>Authentication type</synopsis>
688 Authentication objects hold the authentication information for use
689 by other objects such as <literal>endpoints</literal> or <literal>registrations</literal>.
690 This also allows for multiple objects to use a single auth object. See
691 the <literal>auth_type</literal> config option for password style choices.
692 </para></description>
693 <configOption name="auth_type" default="userpass">
694 <synopsis>Authentication type</synopsis>
696 This option specifies which of the password style config options should be read
697 when trying to authenticate an endpoint inbound request. If set to <literal>userpass</literal>
698 then we'll read from the 'password' option. For <literal>md5</literal> we'll read
703 <enum name="userpass"/>
707 <configOption name="nonce_lifetime" default="32">
708 <synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
710 <configOption name="md5_cred">
711 <synopsis>MD5 Hash used for authentication.</synopsis>
712 <description><para>Only used when auth_type is <literal>md5</literal>.</para></description>
714 <configOption name="password">
715 <synopsis>PlainText password used for authentication.</synopsis>
716 <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
718 <configOption name="realm" default="asterisk">
719 <synopsis>SIP realm for endpoint</synopsis>
721 <configOption name="type">
722 <synopsis>Must be 'auth'</synopsis>
724 <configOption name="username">
725 <synopsis>Username to use for account</synopsis>
728 <configObject name="domain_alias">
729 <synopsis>Domain Alias</synopsis>
731 Signifies that a domain is an alias. If the domain on a session is
732 not found to match an AoR then this object is used to see if we have
733 an alias for the AoR to which the endpoint is binding. This objects
734 name as defined in configuration should be the domain alias and a
735 config option is provided to specify the domain to be aliased.
736 </para></description>
737 <configOption name="type">
738 <synopsis>Must be of type 'domain_alias'.</synopsis>
740 <configOption name="domain">
741 <synopsis>Domain to be aliased</synopsis>
744 <configObject name="transport">
745 <synopsis>SIP Transport</synopsis>
747 <emphasis>Transports</emphasis>
749 <para>There are different transports and protocol derivatives
750 supported by <literal>res_pjsip</literal>. They are in order of
751 preference: UDP, TCP, and WebSocket (WS).</para>
752 <note><para>Changes to transport configuration in pjsip.conf will only be
753 effected on a complete restart of Asterisk. A module reload
754 will not suffice.</para></note>
756 <configOption name="async_operations" default="1">
757 <synopsis>Number of simultaneous Asynchronous Operations</synopsis>
759 <configOption name="bind">
760 <synopsis>IP Address and optional port to bind to for this transport</synopsis>
762 <configOption name="ca_list_file">
763 <synopsis>File containing a list of certificates to read (TLS ONLY)</synopsis>
765 <configOption name="cert_file">
766 <synopsis>Certificate file for endpoint (TLS ONLY)</synopsis>
768 <configOption name="cipher">
769 <synopsis>Preferred Cryptography Cipher (TLS ONLY)</synopsis>
771 Many options for acceptable ciphers see link for more:
772 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
773 </para></description>
775 <configOption name="domain">
776 <synopsis>Domain the transport comes from</synopsis>
778 <configOption name="external_media_address">
779 <synopsis>External IP address to use in RTP handling</synopsis>
781 When a request or response is sent out, if the destination of the
782 message is outside the IP network defined in the option <literal>localnet</literal>,
783 and the media address in the SDP is within the localnet network, then the
784 media address in the SDP will be rewritten to the value defined for
785 <literal>external_media_address</literal>.
786 </para></description>
788 <configOption name="external_signaling_address">
789 <synopsis>External address for SIP signalling</synopsis>
791 <configOption name="external_signaling_port" default="0">
792 <synopsis>External port for SIP signalling</synopsis>
794 <configOption name="method">
795 <synopsis>Method of SSL transport (TLS ONLY)</synopsis>
798 <enum name="default" />
799 <enum name="unspecified" />
800 <enum name="tlsv1" />
801 <enum name="sslv2" />
802 <enum name="sslv3" />
803 <enum name="sslv23" />
807 <configOption name="local_net">
808 <synopsis>Network to consider local (used for NAT purposes).</synopsis>
809 <description><para>This must be in CIDR or dotted decimal format with the IP
810 and mask separated with a slash ('/').</para></description>
812 <configOption name="password">
813 <synopsis>Password required for transport</synopsis>
815 <configOption name="priv_key_file">
816 <synopsis>Private key file (TLS ONLY)</synopsis>
818 <configOption name="protocol" default="udp">
819 <synopsis>Protocol to use for SIP traffic</synopsis>
830 <configOption name="require_client_cert" default="false">
831 <synopsis>Require client certificate (TLS ONLY)</synopsis>
833 <configOption name="type">
834 <synopsis>Must be of type 'transport'.</synopsis>
836 <configOption name="verify_client" default="false">
837 <synopsis>Require verification of client certificate (TLS ONLY)</synopsis>
839 <configOption name="verify_server" default="false">
840 <synopsis>Require verification of server certificate (TLS ONLY)</synopsis>
842 <configOption name="tos" default="false">
843 <synopsis>Enable TOS for the signalling sent over this transport</synopsis>
845 <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
846 for more information on this parameter.</para>
847 <note><para>This option does not apply to the <replaceable>ws</replaceable>
848 or the <replaceable>wss</replaceable> protocols.</para></note>
851 <configOption name="cos" default="false">
852 <synopsis>Enable COS for the signalling sent over this transport</synopsis>
854 <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
855 for more information on this parameter.</para>
856 <note><para>This option does not apply to the <replaceable>ws</replaceable>
857 or the <replaceable>wss</replaceable> protocols.</para></note>
861 <configObject name="contact">
862 <synopsis>A way of creating an aliased name to a SIP URI</synopsis>
864 Contacts are a way to hide SIP URIs from the dialplan directly.
865 They are also used to make a group of contactable parties when
866 in use with <literal>AoR</literal> lists.
867 </para></description>
868 <configOption name="type">
869 <synopsis>Must be of type 'contact'.</synopsis>
871 <configOption name="uri">
872 <synopsis>SIP URI to contact peer</synopsis>
874 <configOption name="expiration_time">
875 <synopsis>Time to keep alive a contact</synopsis>
877 Time to keep alive a contact. String style specification.
878 </para></description>
880 <configOption name="qualify_frequency" default="0">
881 <synopsis>Interval at which to qualify a contact</synopsis>
883 Interval between attempts to qualify the contact for reachability.
884 If <literal>0</literal> never qualify. Time in seconds.
885 </para></description>
887 <configOption name="outbound_proxy">
888 <synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
890 If set the provided URI will be used as the outbound proxy when an
891 OPTIONS request is sent to a contact for qualify purposes.
892 </para></description>
894 <configOption name="path">
895 <synopsis>Stored Path vector for use in Route headers on outgoing requests.</synopsis>
898 <configObject name="aor">
899 <synopsis>The configuration for a location of an endpoint</synopsis>
901 An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
902 AoRs are specified, an endpoint will not be reachable by Asterisk.
903 Beyond that, an AoR has other uses within Asterisk, such as inbound
906 An <literal>AoR</literal> is a way to allow dialing a group
907 of <literal>Contacts</literal> that all use the same
908 <literal>endpoint</literal> for calls.
910 This can be used as another way of grouping a list of contacts to dial
911 rather than specifing them each directly when dialing via the dialplan.
912 This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
914 Registrations: For Asterisk to match an inbound registration to an endpoint,
915 the AoR object name must match the user portion of the SIP URI in the "To:"
916 header of the inbound SIP registration. That will usually be equivalent
917 to the "user name" set in your hard or soft phones configuration.
918 </para></description>
919 <configOption name="contact">
920 <synopsis>Permanent contacts assigned to AoR</synopsis>
922 Contacts specified will be called whenever referenced
923 by <literal>chan_pjsip</literal>.
925 Use a separate "contact=" entry for each contact required. Contacts
926 are specified using a SIP URI.
927 </para></description>
929 <configOption name="default_expiration" default="3600">
930 <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
932 <configOption name="mailboxes">
933 <synopsis>Mailbox(es) to be associated with</synopsis>
934 <description><para>This option applies when an external entity subscribes to an AoR
935 for message waiting indications. The mailboxes specified will be subscribed to.
936 More than one mailbox can be specified with a comma-delimited string.
937 For endpoints that cannot SUBSCRIBE for MWI, you can set the <literal>mailboxes</literal> option in your
938 Endpoint configuration section.
939 </para></description>
941 <configOption name="maximum_expiration" default="7200">
942 <synopsis>Maximum time to keep an AoR</synopsis>
944 Maximium time to keep a peer with explicit expiration. Time in seconds.
945 </para></description>
947 <configOption name="max_contacts" default="0">
948 <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
950 Maximum number of contacts that can associate with this AoR. This value does
951 not affect the number of contacts that can be added with the "contact" option.
952 It only limits contacts added through external interaction, such as
955 <note><para>This should be set to <literal>1</literal> and
956 <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
957 wish to stick with the older <literal>chan_sip</literal> behaviour.
961 <configOption name="minimum_expiration" default="60">
962 <synopsis>Minimum keep alive time for an AoR</synopsis>
964 Minimum time to keep a peer with an explict expiration. Time in seconds.
965 </para></description>
967 <configOption name="remove_existing" default="no">
968 <synopsis>Determines whether new contacts replace existing ones.</synopsis>
970 On receiving a new registration to the AoR should it remove
971 the existing contact that was registered against it?
973 <note><para>This should be set to <literal>yes</literal> and
974 <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
975 wish to stick with the older <literal>chan_sip</literal> behaviour.
979 <configOption name="type">
980 <synopsis>Must be of type 'aor'.</synopsis>
982 <configOption name="qualify_frequency" default="0">
983 <synopsis>Interval at which to qualify an AoR</synopsis>
985 Interval between attempts to qualify the AoR for reachability.
986 If <literal>0</literal> never qualify. Time in seconds.
987 </para></description>
989 <configOption name="authenticate_qualify" default="no">
990 <synopsis>Authenticates a qualify request if needed</synopsis>
992 If true and a qualify request receives a challenge or authenticate response
993 authentication is attempted before declaring the contact available.
994 </para></description>
996 <configOption name="outbound_proxy">
997 <synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
999 If set the provided URI will be used as the outbound proxy when an
1000 OPTIONS request is sent to a contact for qualify purposes.
1001 </para></description>
1003 <configOption name="support_path">
1004 <synopsis>Enables Path support for REGISTER requests and Route support for other requests.</synopsis>
1006 When this option is enabled, the Path headers in register requests will be saved
1007 and its contents will be used in Route headers for outbound out-of-dialog requests
1008 and in Path headers for outbound 200 responses. Path support will also be indicated
1009 in the Supported header.
1010 </para></description>
1013 <configObject name="system">
1014 <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
1016 The settings in this section are global. In addition to being global, the values will
1017 not be re-evaluated when a reload is performed. This is because the values must be set
1018 before the SIP stack is initialized. The only way to reset these values is to either
1019 restart Asterisk, or unload res_pjsip.so and then load it again.
1020 </para></description>
1021 <configOption name="timer_t1" default="500">
1022 <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
1024 Timer T1 is the base for determining how long to wait before retransmitting
1025 requests that receive no response when using an unreliable transport (e.g. UDP).
1026 For more information on this timer, see RFC 3261, Section 17.1.1.1.
1027 </para></description>
1029 <configOption name="timer_b" default="32000">
1030 <synopsis>Set transaction timer B value (milliseconds).</synopsis>
1032 Timer B determines the maximum amount of time to wait after sending an INVITE
1033 request before terminating the transaction. It is recommended that this be set
1034 to 64 * Timer T1, but it may be set higher if desired. For more information on
1035 this timer, see RFC 3261, Section 17.1.1.1.
1036 </para></description>
1038 <configOption name="compact_headers" default="no">
1039 <synopsis>Use the short forms of common SIP header names.</synopsis>
1041 <configOption name="threadpool_initial_size" default="0">
1042 <synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
1044 <configOption name="threadpool_auto_increment" default="5">
1045 <synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
1047 <configOption name="threadpool_idle_timeout" default="60">
1048 <synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
1050 <configOption name="threadpool_max_size" default="0">
1051 <synopsis>Maximum number of threads in the res_pjsip threadpool.
1052 A value of 0 indicates no maximum.</synopsis>
1054 <configOption name="type">
1055 <synopsis>Must be of type 'system'.</synopsis>
1058 <configObject name="global">
1059 <synopsis>Options that apply globally to all SIP communications</synopsis>
1061 The settings in this section are global. Unlike options in the <literal>system</literal>
1062 section, these options can be refreshed by performing a reload.
1063 </para></description>
1064 <configOption name="max_forwards" default="70">
1065 <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
1067 <configOption name="type">
1068 <synopsis>Must be of type 'global'.</synopsis>
1070 <configOption name="user_agent" default="Asterisk <Asterisk Version>">
1071 <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
1073 <configOption name="default_outbound_endpoint" default="default_outbound_endpoint">
1074 <synopsis>Endpoint to use when sending an outbound request to a URI without a specified endpoint.</synopsis>
1080 <manager name="PJSIPQualify" language="en_US">
1082 Qualify a chan_pjsip endpoint.
1085 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1086 <parameter name="Endpoint" required="true">
1087 <para>The endpoint you want to qualify.</para>
1091 <para>Qualify a chan_pjsip endpoint.</para>
1094 <manager name="PJSIPShowEndpoints" language="en_US">
1096 Lists PJSIP endpoints.
1101 Provides a listing of all endpoints. For each endpoint an <literal>EndpointList</literal> event
1102 is raised that contains relevant attributes and status information. Once all
1103 endpoints have been listed an <literal>EndpointListComplete</literal> event is issued.
1107 <manager name="PJSIPShowEndpoint" language="en_US">
1109 Detail listing of an endpoint and its objects.
1112 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1113 <parameter name="Endpoint" required="true">
1114 <para>The endpoint to list.</para>
1119 Provides a detailed listing of options for a given endpoint. Events are issued
1120 showing the configuration and status of the endpoint and associated objects. These
1121 events include <literal>EndpointDetail</literal>, <literal>AorDetail</literal>,
1122 <literal>AuthDetail</literal>, <literal>TransportDetail</literal>, and
1123 <literal>IdentifyDetail</literal>. Some events may be listed multiple times if multiple objects are
1124 associated (for instance AoRs). Once all detail events have been raised a final
1125 <literal>EndpointDetailComplete</literal> event is issued.
1131 #define MOD_DATA_CONTACT "contact"
1133 static pjsip_endpoint *ast_pjsip_endpoint;
1135 static struct ast_threadpool *sip_threadpool;
1137 static int register_service(void *data)
1139 pjsip_module **module = data;
1140 if (!ast_pjsip_endpoint) {
1141 ast_log(LOG_ERROR, "There is no PJSIP endpoint. Unable to register services\n");
1144 if (pjsip_endpt_register_module(ast_pjsip_endpoint, *module) != PJ_SUCCESS) {
1145 ast_log(LOG_ERROR, "Unable to register module %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1148 ast_debug(1, "Registered SIP service %.*s (%p)\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name), *module);
1149 ast_module_ref(ast_module_info->self);
1153 int ast_sip_register_service(pjsip_module *module)
1155 return ast_sip_push_task_synchronous(NULL, register_service, &module);
1158 static int unregister_service(void *data)
1160 pjsip_module **module = data;
1161 ast_module_unref(ast_module_info->self);
1162 if (!ast_pjsip_endpoint) {
1165 pjsip_endpt_unregister_module(ast_pjsip_endpoint, *module);
1166 ast_debug(1, "Unregistered SIP service %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1170 void ast_sip_unregister_service(pjsip_module *module)
1172 ast_sip_push_task_synchronous(NULL, unregister_service, &module);
1175 static struct ast_sip_authenticator *registered_authenticator;
1177 int ast_sip_register_authenticator(struct ast_sip_authenticator *auth)
1179 if (registered_authenticator) {
1180 ast_log(LOG_WARNING, "Authenticator %p is already registered. Cannot register a new one\n", registered_authenticator);
1183 registered_authenticator = auth;
1184 ast_debug(1, "Registered SIP authenticator module %p\n", auth);
1185 ast_module_ref(ast_module_info->self);
1189 void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth)
1191 if (registered_authenticator != auth) {
1192 ast_log(LOG_WARNING, "Trying to unregister authenticator %p but authenticator %p registered\n",
1193 auth, registered_authenticator);
1196 registered_authenticator = NULL;
1197 ast_debug(1, "Unregistered SIP authenticator %p\n", auth);
1198 ast_module_unref(ast_module_info->self);
1201 int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1203 if (!registered_authenticator) {
1204 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is not required\n");
1208 return registered_authenticator->requires_authentication(endpoint, rdata);
1211 enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
1212 pjsip_rx_data *rdata, pjsip_tx_data *tdata)
1214 if (!registered_authenticator) {
1215 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is successful\n");
1218 return registered_authenticator->check_authentication(endpoint, rdata, tdata);
1221 static struct ast_sip_outbound_authenticator *registered_outbound_authenticator;
1223 int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1225 if (registered_outbound_authenticator) {
1226 ast_log(LOG_WARNING, "Outbound authenticator %p is already registered. Cannot register a new one\n", registered_outbound_authenticator);
1229 registered_outbound_authenticator = auth;
1230 ast_debug(1, "Registered SIP outbound authenticator module %p\n", auth);
1231 ast_module_ref(ast_module_info->self);
1235 void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1237 if (registered_outbound_authenticator != auth) {
1238 ast_log(LOG_WARNING, "Trying to unregister outbound authenticator %p but outbound authenticator %p registered\n",
1239 auth, registered_outbound_authenticator);
1242 registered_outbound_authenticator = NULL;
1243 ast_debug(1, "Unregistered SIP outbound authenticator %p\n", auth);
1244 ast_module_unref(ast_module_info->self);
1247 int ast_sip_create_request_with_auth(const struct ast_sip_auth_vector *auths, pjsip_rx_data *challenge,
1248 pjsip_transaction *tsx, pjsip_tx_data **new_request)
1250 if (!registered_outbound_authenticator) {
1251 ast_log(LOG_WARNING, "No SIP outbound authenticator registered. Cannot respond to authentication challenge\n");
1254 return registered_outbound_authenticator->create_request_with_auth(auths, challenge, tsx, new_request);
1257 struct endpoint_identifier_list {
1258 struct ast_sip_endpoint_identifier *identifier;
1259 AST_RWLIST_ENTRY(endpoint_identifier_list) list;
1262 static AST_RWLIST_HEAD_STATIC(endpoint_identifiers, endpoint_identifier_list);
1264 int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1266 struct endpoint_identifier_list *id_list_item;
1267 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1269 id_list_item = ast_calloc(1, sizeof(*id_list_item));
1270 if (!id_list_item) {
1271 ast_log(LOG_ERROR, "Unabled to add endpoint identifier. Out of memory.\n");
1274 id_list_item->identifier = identifier;
1276 AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list);
1277 ast_debug(1, "Registered endpoint identifier %p\n", identifier);
1279 ast_module_ref(ast_module_info->self);
1283 void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1285 struct endpoint_identifier_list *iter;
1286 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1287 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) {
1288 if (iter->identifier == identifier) {
1289 AST_RWLIST_REMOVE_CURRENT(list);
1291 ast_debug(1, "Unregistered endpoint identifier %p\n", identifier);
1292 ast_module_unref(ast_module_info->self);
1296 AST_RWLIST_TRAVERSE_SAFE_END;
1299 struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata)
1301 struct endpoint_identifier_list *iter;
1302 struct ast_sip_endpoint *endpoint = NULL;
1303 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1304 AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) {
1305 ast_assert(iter->identifier->identify_endpoint != NULL);
1306 endpoint = iter->identifier->identify_endpoint(rdata);
1314 AST_RWLIST_HEAD_STATIC(endpoint_formatters, ast_sip_endpoint_formatter);
1316 int ast_sip_register_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
1318 SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1319 AST_RWLIST_INSERT_TAIL(&endpoint_formatters, obj, next);
1320 ast_module_ref(ast_module_info->self);
1324 void ast_sip_unregister_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
1326 struct ast_sip_endpoint_formatter *i;
1327 SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1328 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_formatters, i, next) {
1330 AST_RWLIST_REMOVE_CURRENT(next);
1331 ast_module_unref(ast_module_info->self);
1335 AST_RWLIST_TRAVERSE_SAFE_END;
1338 int ast_sip_format_endpoint_ami(struct ast_sip_endpoint *endpoint,
1339 struct ast_sip_ami *ami, int *count)
1342 struct ast_sip_endpoint_formatter *i;
1343 SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1345 AST_RWLIST_TRAVERSE(&endpoint_formatters, i, next) {
1346 if (i->format_ami && ((res = i->format_ami(endpoint, ami)) < 0)) {
1357 pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
1359 return ast_pjsip_endpoint;
1362 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)
1364 pj_str_t tmp, local_addr;
1366 pjsip_sip_uri *sip_uri;
1367 pjsip_transport_type_e type = PJSIP_TRANSPORT_UNSPECIFIED;
1369 char uuid_str[AST_UUID_STR_LEN];
1371 if (ast_strlen_zero(user)) {
1372 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1376 user = ast_uuid_to_str(uuid, uuid_str, sizeof(uuid_str));
1379 /* Parse the provided target URI so we can determine what transport it will end up using */
1380 pj_strdup_with_null(pool, &tmp, target);
1382 if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) ||
1383 (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
1387 sip_uri = pjsip_uri_get_uri(uri);
1389 /* Determine the transport type to use */
1390 if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
1391 type = PJSIP_TRANSPORT_TLS;
1392 } else if (!sip_uri->transport_param.slen) {
1393 type = PJSIP_TRANSPORT_UDP;
1395 type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
1398 if (type == PJSIP_TRANSPORT_UNSPECIFIED) {
1402 /* If the host is IPv6 turn the transport into an IPv6 version */
1403 if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1404 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1407 if (!ast_strlen_zero(domain)) {
1408 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1409 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1413 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1414 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1418 /* Get the local bound address for the transport that will be used when communicating with the provided URI */
1419 if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), pool, type, selector,
1420 &local_addr, &local_port) != PJ_SUCCESS) {
1422 /* If no local address can be retrieved using the transport manager use the host one */
1423 pj_strdup(pool, &local_addr, pj_gethostname());
1424 local_port = pjsip_transport_get_default_port_for_type(PJSIP_TRANSPORT_UDP);
1427 /* If IPv6 was specified in the transport, set the proper type */
1428 if (pj_strchr(&local_addr, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1429 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1432 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1433 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1434 "<sip:%s@%s%.*s%s:%d%s%s>",
1436 (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1437 (int)local_addr.slen,
1439 (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1441 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1442 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1447 static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
1449 RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
1450 const char *transport_name = endpoint->transport;
1452 if (ast_strlen_zero(transport_name)) {
1456 transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name);
1458 if (!transport || !transport->state) {
1459 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport '%s' for endpoint '%s'\n",
1460 transport_name, ast_sorcery_object_get_id(endpoint));
1464 if (transport->state->transport) {
1465 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1466 selector->u.transport = transport->state->transport;
1467 } else if (transport->state->factory) {
1468 selector->type = PJSIP_TPSELECTOR_LISTENER;
1469 selector->u.listener = transport->state->factory;
1477 pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
1479 char enclosed_uri[PJSIP_MAX_URL_SIZE];
1480 pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri, target_uri;
1481 pjsip_dialog *dlg = NULL;
1482 const char *outbound_proxy = endpoint->outbound_proxy;
1483 pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1484 static const pj_str_t HCONTACT = { "Contact", 7 };
1486 snprintf(enclosed_uri, sizeof(enclosed_uri), "<%s>", uri);
1487 pj_cstr(&remote_uri, enclosed_uri);
1489 pj_cstr(&target_uri, uri);
1491 if (pjsip_dlg_create_uac(pjsip_ua_instance(), &local_uri, NULL, &remote_uri, &target_uri, &dlg) != PJ_SUCCESS) {
1495 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1496 pjsip_dlg_terminate(dlg);
1500 if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
1501 pjsip_dlg_terminate(dlg);
1505 /* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
1506 pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
1507 dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
1508 dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
1510 /* If a request user has been specified and we are permitted to change it, do so */
1511 if (!ast_strlen_zero(request_user)) {
1512 pjsip_sip_uri *sip_uri;
1514 if (PJSIP_URI_SCHEME_IS_SIP(dlg->target) || PJSIP_URI_SCHEME_IS_SIPS(dlg->target)) {
1515 sip_uri = pjsip_uri_get_uri(dlg->target);
1516 pj_strdup2(dlg->pool, &sip_uri->user, request_user);
1518 if (PJSIP_URI_SCHEME_IS_SIP(dlg->remote.info->uri) || PJSIP_URI_SCHEME_IS_SIPS(dlg->remote.info->uri)) {
1519 sip_uri = pjsip_uri_get_uri(dlg->remote.info->uri);
1520 pj_strdup2(dlg->pool, &sip_uri->user, request_user);
1524 /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
1527 pjsip_dlg_set_transport(dlg, &selector);
1529 if (!ast_strlen_zero(outbound_proxy)) {
1530 pjsip_route_hdr route_set, *route;
1531 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1534 pj_list_init(&route_set);
1536 pj_strdup2_with_null(dlg->pool, &tmp, outbound_proxy);
1537 if (!(route = pjsip_parse_hdr(dlg->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1539 pjsip_dlg_terminate(dlg);
1542 pj_list_push_back(&route_set, route);
1544 pjsip_dlg_set_route_set(dlg, &route_set);
1552 pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1556 pjsip_transport_type_e type = rdata->tp_info.transport->key.type;
1559 contact.ptr = pj_pool_alloc(rdata->tp_info.pool, PJSIP_MAX_URL_SIZE);
1560 contact.slen = pj_ansi_snprintf(contact.ptr, PJSIP_MAX_URL_SIZE,
1561 "<sip:%s%.*s%s:%d%s%s>",
1562 (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1563 (int)rdata->tp_info.transport->local_name.host.slen,
1564 rdata->tp_info.transport->local_name.host.ptr,
1565 (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1566 rdata->tp_info.transport->local_name.port,
1567 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1568 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1570 status = pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, &contact, &dlg);
1571 if (status != PJ_SUCCESS) {
1572 char err[PJ_ERR_MSG_SIZE];
1574 pj_strerror(status, err, sizeof(err));
1575 ast_log(LOG_ERROR, "Could not create dialog with endpoint %s. %s\n",
1576 ast_sorcery_object_get_id(endpoint), err);
1583 /* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
1584 static const pjsip_method info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
1585 static const pjsip_method message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
1589 const pjsip_method *pmethod;
1591 { "INVITE", &pjsip_invite_method },
1592 { "CANCEL", &pjsip_cancel_method },
1593 { "ACK", &pjsip_ack_method },
1594 { "BYE", &pjsip_bye_method },
1595 { "REGISTER", &pjsip_register_method },
1596 { "OPTIONS", &pjsip_options_method },
1597 { "SUBSCRIBE", &pjsip_subscribe_method },
1598 { "NOTIFY", &pjsip_notify_method },
1599 { "PUBLISH", &pjsip_publish_method },
1600 { "INFO", &info_method },
1601 { "MESSAGE", &message_method },
1604 static const pjsip_method *get_pjsip_method(const char *method)
1607 for (i = 0; i < ARRAY_LEN(methods); ++i) {
1608 if (!strcmp(method, methods[i].method)) {
1609 return methods[i].pmethod;
1615 static int create_in_dialog_request(const pjsip_method *method, struct pjsip_dialog *dlg, pjsip_tx_data **tdata)
1617 if (pjsip_dlg_create_request(dlg, method, -1, tdata) != PJ_SUCCESS) {
1618 ast_log(LOG_WARNING, "Unable to create in-dialog request.\n");
1625 static pj_bool_t supplement_on_rx_request(pjsip_rx_data *rdata);
1626 static pjsip_module supplement_module = {
1627 .name = { "Out of dialog supplement hook", 29 },
1629 .priority = PJSIP_MOD_PRIORITY_APPLICATION - 1,
1630 .on_rx_request = supplement_on_rx_request,
1633 static int create_out_of_dialog_request(const pjsip_method *method, struct ast_sip_endpoint *endpoint,
1634 const char *uri, struct ast_sip_contact *provided_contact, pjsip_tx_data **tdata)
1636 RAII_VAR(struct ast_sip_contact *, contact, ao2_bump(provided_contact), ao2_cleanup);
1637 pj_str_t remote_uri;
1640 pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1642 if (ast_strlen_zero(uri)) {
1643 if (!endpoint && !contact) {
1644 ast_log(LOG_ERROR, "An endpoint and/or uri must be specified\n");
1649 contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
1651 if (!contact || ast_strlen_zero(contact->uri)) {
1652 ast_log(LOG_ERROR, "Unable to retrieve contact for endpoint %s\n",
1653 ast_sorcery_object_get_id(endpoint));
1657 pj_cstr(&remote_uri, contact->uri);
1659 pj_cstr(&remote_uri, uri);
1663 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1664 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
1665 ast_sorcery_object_get_id(endpoint));
1670 pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
1673 ast_log(LOG_ERROR, "Unable to create PJLIB memory pool\n");
1677 if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL,
1678 endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
1679 ast_log(LOG_ERROR, "Unable to create From header for %.*s request to endpoint %s\n",
1680 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1681 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1685 if (pjsip_endpt_create_request(ast_sip_get_pjsip_endpoint(), method, &remote_uri,
1686 &from, &remote_uri, &from, NULL, -1, NULL, tdata) != PJ_SUCCESS) {
1687 ast_log(LOG_ERROR, "Unable to create outbound %.*s request to endpoint %s\n",
1688 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1689 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1693 /* If an outbound proxy is specified on the endpoint apply it to this request */
1694 if (endpoint && !ast_strlen_zero(endpoint->outbound_proxy) &&
1695 ast_sip_set_outbound_proxy((*tdata), endpoint->outbound_proxy)) {
1696 ast_log(LOG_ERROR, "Unable to apply outbound proxy on request %.*s to endpoint %s\n",
1697 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1698 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1702 ast_sip_mod_data_set((*tdata)->pool, (*tdata)->mod_data, supplement_module.id, MOD_DATA_CONTACT, ao2_bump(contact));
1704 /* We can release this pool since request creation copied all the necessary
1705 * data into the outbound request's pool
1707 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1711 int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
1712 struct ast_sip_endpoint *endpoint, const char *uri,
1713 struct ast_sip_contact *contact, pjsip_tx_data **tdata)
1715 const pjsip_method *pmethod = get_pjsip_method(method);
1718 ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
1723 return create_in_dialog_request(pmethod, dlg, tdata);
1725 return create_out_of_dialog_request(pmethod, endpoint, uri, contact, tdata);
1729 AST_RWLIST_HEAD_STATIC(supplements, ast_sip_supplement);
1731 int ast_sip_register_supplement(struct ast_sip_supplement *supplement)
1733 struct ast_sip_supplement *iter;
1735 SCOPED_LOCK(lock, &supplements, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1737 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&supplements, iter, next) {
1738 if (iter->priority > supplement->priority) {
1739 AST_RWLIST_INSERT_BEFORE_CURRENT(supplement, next);
1744 AST_RWLIST_TRAVERSE_SAFE_END;
1747 AST_RWLIST_INSERT_TAIL(&supplements, supplement, next);
1749 ast_module_ref(ast_module_info->self);
1753 void ast_sip_unregister_supplement(struct ast_sip_supplement *supplement)
1755 struct ast_sip_supplement *iter;
1756 SCOPED_LOCK(lock, &supplements, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1757 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&supplements, iter, next) {
1758 if (supplement == iter) {
1759 AST_RWLIST_REMOVE_CURRENT(next);
1760 ast_module_unref(ast_module_info->self);
1764 AST_RWLIST_TRAVERSE_SAFE_END;
1767 static int send_in_dialog_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg)
1769 if (pjsip_dlg_send_request(dlg, tdata, -1, NULL) != PJ_SUCCESS) {
1770 ast_log(LOG_WARNING, "Unable to send in-dialog request.\n");
1776 static pj_bool_t does_method_match(const pj_str_t *message_method, const char *supplement_method)
1780 if (ast_strlen_zero(supplement_method)) {
1784 pj_cstr(&method, supplement_method);
1786 return pj_stristr(&method, message_method) ? PJ_TRUE : PJ_FALSE;
1789 /*! \brief Structure to hold information about an outbound request */
1790 struct send_request_data {
1791 struct ast_sip_endpoint *endpoint; /*! The endpoint associated with this request */
1792 void *token; /*! Information to be provided to the callback upon receipt of a response */
1793 void (*callback)(void *token, pjsip_event *e); /*! The callback to be called upon receipt of a response */
1796 static void send_request_data_destroy(void *obj)
1798 struct send_request_data *req_data = obj;
1799 ao2_cleanup(req_data->endpoint);
1802 static struct send_request_data *send_request_data_alloc(struct ast_sip_endpoint *endpoint,
1803 void *token, void (*callback)(void *token, pjsip_event *e))
1805 struct send_request_data *req_data = ao2_alloc(sizeof(*req_data), send_request_data_destroy);
1811 req_data->endpoint = ao2_bump(endpoint);
1812 req_data->token = token;
1813 req_data->callback = callback;
1818 static void send_request_cb(void *token, pjsip_event *e)
1820 RAII_VAR(struct send_request_data *, req_data, token, ao2_cleanup);
1821 pjsip_transaction *tsx = e->body.tsx_state.tsx;
1822 pjsip_rx_data *challenge = e->body.tsx_state.src.rdata;
1823 pjsip_tx_data *tdata;
1824 struct ast_sip_supplement *supplement;
1826 AST_RWLIST_RDLOCK(&supplements);
1827 AST_LIST_TRAVERSE(&supplements, supplement, next) {
1828 if (supplement->incoming_response && does_method_match(&challenge->msg_info.cseq->method.name, supplement->method)) {
1829 supplement->incoming_response(req_data->endpoint, challenge);
1832 AST_RWLIST_UNLOCK(&supplements);
1834 if (tsx->status_code == 401 || tsx->status_code == 407) {
1835 if (!ast_sip_create_request_with_auth(&req_data->endpoint->outbound_auths, challenge, tsx, &tdata)) {
1836 pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, req_data->token, req_data->callback);
1841 if (req_data->callback) {
1842 req_data->callback(req_data->token, e);
1846 static int send_out_of_dialog_request(pjsip_tx_data *tdata, struct ast_sip_endpoint *endpoint,
1847 void *token, void (*callback)(void *token, pjsip_event *e))
1849 struct ast_sip_supplement *supplement;
1850 struct send_request_data *req_data = send_request_data_alloc(endpoint, token, callback);
1851 struct ast_sip_contact *contact = ast_sip_mod_data_get(tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT);
1857 AST_RWLIST_RDLOCK(&supplements);
1858 AST_LIST_TRAVERSE(&supplements, supplement, next) {
1859 if (supplement->outgoing_request && does_method_match(&tdata->msg->line.req.method.name, supplement->method)) {
1860 supplement->outgoing_request(endpoint, contact, tdata);
1863 AST_RWLIST_UNLOCK(&supplements);
1865 ast_sip_mod_data_set(tdata->pool, tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT, NULL);
1866 ao2_cleanup(contact);
1868 if (pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, req_data, send_request_cb) != PJ_SUCCESS) {
1869 ast_log(LOG_ERROR, "Error attempting to send outbound %.*s request to endpoint %s\n",
1870 (int) pj_strlen(&tdata->msg->line.req.method.name),
1871 pj_strbuf(&tdata->msg->line.req.method.name),
1872 ast_sorcery_object_get_id(endpoint));
1873 ao2_cleanup(req_data);
1880 int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg,
1881 struct ast_sip_endpoint *endpoint, void *token,
1882 void (*callback)(void *token, pjsip_event *e))
1884 ast_assert(tdata->msg->type == PJSIP_REQUEST_MSG);
1887 return send_in_dialog_request(tdata, dlg);
1889 return send_out_of_dialog_request(tdata, endpoint, token, callback);
1893 int ast_sip_set_outbound_proxy(pjsip_tx_data *tdata, const char *proxy)
1895 pjsip_route_hdr *route;
1896 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1899 pj_strdup2_with_null(tdata->pool, &tmp, proxy);
1900 if (!(route = pjsip_parse_hdr(tdata->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1904 pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr*)route);
1909 int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value)
1913 pjsip_generic_string_hdr *hdr;
1915 pj_cstr(&hdr_name, name);
1916 pj_cstr(&hdr_value, value);
1918 hdr = pjsip_generic_string_hdr_create(tdata->pool, &hdr_name, &hdr_value);
1920 pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) hdr);
1924 static pjsip_msg_body *ast_body_to_pjsip_body(pj_pool_t *pool, const struct ast_sip_body *body)
1930 pj_cstr(&type, body->type);
1931 pj_cstr(&subtype, body->subtype);
1932 pj_cstr(&body_text, body->body_text);
1934 return pjsip_msg_body_create(pool, &type, &subtype, &body_text);
1937 int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body)
1939 pjsip_msg_body *pjsip_body = ast_body_to_pjsip_body(tdata->pool, body);
1940 tdata->msg->body = pjsip_body;
1944 int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies)
1947 /* NULL for type and subtype automatically creates "multipart/mixed" */
1948 pjsip_msg_body *body = pjsip_multipart_create(tdata->pool, NULL, NULL);
1950 for (i = 0; i < num_bodies; ++i) {
1951 pjsip_multipart_part *part = pjsip_multipart_create_part(tdata->pool);
1952 part->body = ast_body_to_pjsip_body(tdata->pool, bodies[i]);
1953 pjsip_multipart_add_part(tdata->pool, body, part);
1956 tdata->msg->body = body;
1960 int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text)
1962 size_t combined_size = strlen(body_text) + tdata->msg->body->len;
1963 struct ast_str *body_buffer = ast_str_alloca(combined_size);
1965 ast_str_set(&body_buffer, 0, "%.*s%s", (int) tdata->msg->body->len, (char *) tdata->msg->body->data, body_text);
1967 tdata->msg->body->data = pj_pool_alloc(tdata->pool, combined_size);
1968 pj_memcpy(tdata->msg->body->data, ast_str_buffer(body_buffer), combined_size);
1969 tdata->msg->body->len = combined_size;
1974 struct ast_taskprocessor *ast_sip_create_serializer(void)
1976 struct ast_taskprocessor *serializer;
1977 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1978 char name[AST_UUID_STR_LEN];
1984 ast_uuid_to_str(uuid, name, sizeof(name));
1986 serializer = ast_threadpool_serializer(name, sip_threadpool);
1993 int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1996 return ast_taskprocessor_push(serializer, sip_task, task_data);
1998 return ast_threadpool_push(sip_threadpool, sip_task, task_data);
2002 struct sync_task_data {
2007 int (*task)(void *);
2011 static int sync_task(void *data)
2013 struct sync_task_data *std = data;
2014 std->fail = std->task(std->task_data);
2016 ast_mutex_lock(&std->lock);
2018 ast_cond_signal(&std->cond);
2019 ast_mutex_unlock(&std->lock);
2023 int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
2025 /* This method is an onion */
2026 struct sync_task_data std;
2028 if (ast_sip_thread_is_servant()) {
2029 return sip_task(task_data);
2032 ast_mutex_init(&std.lock);
2033 ast_cond_init(&std.cond, NULL);
2034 std.fail = std.complete = 0;
2035 std.task = sip_task;
2036 std.task_data = task_data;
2039 if (ast_taskprocessor_push(serializer, sync_task, &std)) {
2043 if (ast_threadpool_push(sip_threadpool, sync_task, &std)) {
2048 ast_mutex_lock(&std.lock);
2049 while (!std.complete) {
2050 ast_cond_wait(&std.cond, &std.lock);
2052 ast_mutex_unlock(&std.lock);
2054 ast_mutex_destroy(&std.lock);
2055 ast_cond_destroy(&std.cond);
2059 void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
2061 size_t chars_to_copy = MIN(size - 1, pj_strlen(src));
2062 memcpy(dest, pj_strbuf(src), chars_to_copy);
2063 dest[chars_to_copy] = '\0';
2066 int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype)
2068 pjsip_media_type compare;
2070 if (!content_type) {
2074 pjsip_media_type_init2(&compare, type, subtype);
2076 return pjsip_media_type_cmp(content_type, &compare, 0) ? 0 : -1;
2079 pj_caching_pool caching_pool;
2080 pj_pool_t *memory_pool;
2081 pj_thread_t *monitor_thread;
2082 static int monitor_continue;
2084 static void *monitor_thread_exec(void *endpt)
2086 while (monitor_continue) {
2087 const pj_time_val delay = {0, 10};
2088 pjsip_endpt_handle_events(ast_pjsip_endpoint, &delay);
2093 static void stop_monitor_thread(void)
2095 monitor_continue = 0;
2096 pj_thread_join(monitor_thread);
2099 AST_THREADSTORAGE(pj_thread_storage);
2100 AST_THREADSTORAGE(servant_id_storage);
2101 #define SIP_SERVANT_ID 0x5E2F1D
2103 static void sip_thread_start(void)
2105 pj_thread_desc *desc;
2106 pj_thread_t *thread;
2107 uint32_t *servant_id;
2109 servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
2111 ast_log(LOG_ERROR, "Could not set SIP servant ID in thread-local storage.\n");
2114 *servant_id = SIP_SERVANT_ID;
2116 desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc));
2118 ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n");
2121 pj_bzero(*desc, sizeof(*desc));
2123 if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) {
2124 ast_log(LOG_ERROR, "Couldn't register thread with PJLIB.\n");
2128 int ast_sip_thread_is_servant(void)
2130 uint32_t *servant_id;
2132 servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
2137 return *servant_id == SIP_SERVANT_ID;
2140 void *ast_sip_dict_get(void *ht, const char *key)
2142 unsigned int hval = 0;
2148 return pj_hash_get(ht, key, PJ_HASH_KEY_STRING, &hval);
2151 void *ast_sip_dict_set(pj_pool_t* pool, void *ht,
2152 const char *key, void *val)
2155 ht = pj_hash_create(pool, 11);
2158 pj_hash_set(pool, ht, key, PJ_HASH_KEY_STRING, 0, val);
2163 static pj_bool_t supplement_on_rx_request(pjsip_rx_data *rdata)
2165 struct ast_sip_supplement *supplement;
2167 if (pjsip_rdata_get_dlg(rdata)) {
2171 AST_RWLIST_RDLOCK(&supplements);
2172 AST_LIST_TRAVERSE(&supplements, supplement, next) {
2173 if (supplement->incoming_request && does_method_match(&rdata->msg_info.msg->line.req.method.name, supplement->method)) {
2174 supplement->incoming_request(ast_pjsip_rdata_get_endpoint(rdata), rdata);
2177 AST_RWLIST_UNLOCK(&supplements);
2182 int ast_sip_send_response(pjsip_response_addr *res_addr, pjsip_tx_data *tdata, struct ast_sip_endpoint *sip_endpoint)
2184 struct ast_sip_supplement *supplement;
2185 pjsip_cseq_hdr *cseq = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_CSEQ, NULL);
2186 struct ast_sip_contact *contact = ast_sip_mod_data_get(tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT);
2188 AST_RWLIST_RDLOCK(&supplements);
2189 AST_LIST_TRAVERSE(&supplements, supplement, next) {
2190 if (supplement->outgoing_response && does_method_match(&cseq->method.name, supplement->method)) {
2191 supplement->outgoing_response(sip_endpoint, contact, tdata);
2194 AST_RWLIST_UNLOCK(&supplements);
2196 ast_sip_mod_data_set(tdata->pool, tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT, NULL);
2197 ao2_cleanup(contact);
2199 return pjsip_endpt_send_response(ast_sip_get_pjsip_endpoint(), res_addr, tdata, NULL, NULL);
2202 int ast_sip_create_response(const pjsip_rx_data *rdata, int st_code,
2203 struct ast_sip_contact *contact, pjsip_tx_data **tdata)
2205 int res = pjsip_endpt_create_response(ast_sip_get_pjsip_endpoint(), rdata, st_code, NULL, tdata);
2208 ast_sip_mod_data_set((*tdata)->pool, (*tdata)->mod_data, supplement_module.id, MOD_DATA_CONTACT, ao2_bump(contact));
2214 static void remove_request_headers(pjsip_endpoint *endpt)
2216 const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt);
2217 pjsip_hdr *iter = request_headers->next;
2219 while (iter != request_headers) {
2220 pjsip_hdr *to_erase = iter;
2222 pj_list_erase(to_erase);
2226 static int load_module(void)
2228 /* The third parameter is just copied from
2229 * example code from PJLIB. This can be adjusted
2233 struct ast_threadpool_options options;
2235 if (pj_init() != PJ_SUCCESS) {
2236 return AST_MODULE_LOAD_DECLINE;
2239 if (pjlib_util_init() != PJ_SUCCESS) {
2241 return AST_MODULE_LOAD_DECLINE;
2244 pj_caching_pool_init(&caching_pool, NULL, 1024 * 1024);
2245 if (pjsip_endpt_create(&caching_pool.factory, "SIP", &ast_pjsip_endpoint) != PJ_SUCCESS) {
2246 ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n");
2247 pj_caching_pool_destroy(&caching_pool);
2248 return AST_MODULE_LOAD_DECLINE;
2251 /* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that,
2252 * we need to stop PJSIP from doing it automatically
2254 remove_request_headers(ast_pjsip_endpoint);
2256 memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL);
2258 ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n");
2259 pjsip_endpt_destroy(ast_pjsip_endpoint);
2260 ast_pjsip_endpoint = NULL;
2261 pj_caching_pool_destroy(&caching_pool);
2262 return AST_MODULE_LOAD_DECLINE;
2265 if (ast_sip_initialize_system()) {
2266 ast_log(LOG_ERROR, "Failed to initialize SIP 'system' configuration section. Aborting load\n");
2267 pj_pool_release(memory_pool);
2269 pjsip_endpt_destroy(ast_pjsip_endpoint);
2270 ast_pjsip_endpoint = NULL;
2271 pj_caching_pool_destroy(&caching_pool);
2272 return AST_MODULE_LOAD_DECLINE;
2275 sip_get_threadpool_options(&options);
2276 options.thread_start = sip_thread_start;
2277 sip_threadpool = ast_threadpool_create("SIP", NULL, &options);
2278 if (!sip_threadpool) {
2279 ast_log(LOG_ERROR, "Failed to create SIP threadpool. Aborting load\n");
2280 pj_pool_release(memory_pool);
2282 pjsip_endpt_destroy(ast_pjsip_endpoint);
2283 ast_pjsip_endpoint = NULL;
2284 pj_caching_pool_destroy(&caching_pool);
2285 return AST_MODULE_LOAD_DECLINE;
2288 pjsip_tsx_layer_init_module(ast_pjsip_endpoint);
2289 pjsip_ua_init_module(ast_pjsip_endpoint, NULL);
2291 monitor_continue = 1;
2292 status = pj_thread_create(memory_pool, "SIP", (pj_thread_proc *) &monitor_thread_exec,
2293 NULL, PJ_THREAD_DEFAULT_STACK_SIZE * 2, 0, &monitor_thread);
2294 if (status != PJ_SUCCESS) {
2295 ast_log(LOG_ERROR, "Failed to start SIP monitor thread. Aborting load\n");
2296 pj_pool_release(memory_pool);
2298 pjsip_endpt_destroy(ast_pjsip_endpoint);
2299 ast_pjsip_endpoint = NULL;
2300 pj_caching_pool_destroy(&caching_pool);
2301 return AST_MODULE_LOAD_DECLINE;
2304 ast_sip_initialize_global_headers();
2306 if (ast_res_pjsip_initialize_configuration(ast_module_info)) {
2307 ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
2308 ast_sip_destroy_global_headers();
2309 stop_monitor_thread();
2310 pj_pool_release(memory_pool);
2312 pjsip_endpt_destroy(ast_pjsip_endpoint);
2313 ast_pjsip_endpoint = NULL;
2314 pj_caching_pool_destroy(&caching_pool);
2315 return AST_MODULE_LOAD_DECLINE;
2318 if (ast_sip_initialize_distributor()) {
2319 ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n");
2320 ast_res_pjsip_destroy_configuration();
2321 ast_sip_destroy_global_headers();
2322 stop_monitor_thread();
2323 pj_pool_release(memory_pool);
2325 pjsip_endpt_destroy(ast_pjsip_endpoint);
2326 ast_pjsip_endpoint = NULL;
2327 pj_caching_pool_destroy(&caching_pool);
2328 return AST_MODULE_LOAD_DECLINE;
2331 if (ast_sip_register_service(&supplement_module)) {
2332 ast_log(LOG_ERROR, "Failed to initialize supplement hooks. Aborting load\n");
2333 ast_sip_destroy_distributor();
2334 ast_res_pjsip_destroy_configuration();
2335 ast_sip_destroy_global_headers();
2336 stop_monitor_thread();
2337 pj_pool_release(memory_pool);
2339 pjsip_endpt_destroy(ast_pjsip_endpoint);
2340 ast_pjsip_endpoint = NULL;
2341 pj_caching_pool_destroy(&caching_pool);
2342 return AST_MODULE_LOAD_DECLINE;
2345 if (ast_sip_initialize_outbound_authentication()) {
2346 ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n");
2347 ast_sip_unregister_service(&supplement_module);
2348 ast_sip_destroy_distributor();
2349 ast_res_pjsip_destroy_configuration();
2350 ast_sip_destroy_global_headers();
2351 stop_monitor_thread();
2352 pj_pool_release(memory_pool);
2354 pjsip_endpt_destroy(ast_pjsip_endpoint);
2355 ast_pjsip_endpoint = NULL;
2356 pj_caching_pool_destroy(&caching_pool);
2357 return AST_MODULE_LOAD_DECLINE;
2360 ast_res_pjsip_init_options_handling(0);
2362 ast_module_ref(ast_module_info->self);
2364 return AST_MODULE_LOAD_SUCCESS;
2367 static int reload_module(void)
2369 if (ast_res_pjsip_reload_configuration()) {
2370 return AST_MODULE_LOAD_DECLINE;
2372 ast_res_pjsip_init_options_handling(1);
2376 static int unload_module(void)
2378 /* This will never get called as this module can't be unloaded */
2382 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Basic SIP resource",
2383 .load = load_module,
2384 .unload = unload_module,
2385 .reload = reload_module,
2386 .load_pri = AST_MODPRI_CHANNEL_DEPEND - 5,
projects / asterisk/asterisk.git / blob