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. app_voicemail mailboxes must be specified
274 as mailbox@context; for example: mailboxes=6001@default. For mailboxes provided by
275 external sources, such as through the res_external_mwi module, you must specify
276 strings supported by the external system.
278 For endpoints that SUBSCRIBE for MWI, use the <literal>mailboxes</literal> option in your AOR
280 </para></description>
282 <configOption name="moh_suggest" default="default">
283 <synopsis>Default Music On Hold class</synopsis>
285 <configOption name="outbound_auth">
286 <synopsis>Authentication object used for outbound requests</synopsis>
288 <configOption name="outbound_proxy">
289 <synopsis>Proxy through which to send requests, a full SIP URI must be provided</synopsis>
291 <configOption name="rewrite_contact">
292 <synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
294 On inbound SIP messages from this endpoint, the Contact header will be changed to have the
295 source IP address and port. This option does not affect outbound messages send to this
297 </para></description>
299 <configOption name="rtp_ipv6" default="no">
300 <synopsis>Allow use of IPv6 for RTP traffic</synopsis>
302 <configOption name="rtp_symmetric" default="no">
303 <synopsis>Enforce that RTP must be symmetric</synopsis>
305 <configOption name="send_diversion" default="yes">
306 <synopsis>Send the Diversion header, conveying the diversion
307 information to the called user agent</synopsis>
309 <configOption name="send_pai" default="no">
310 <synopsis>Send the P-Asserted-Identity header</synopsis>
312 <configOption name="send_rpid" default="no">
313 <synopsis>Send the Remote-Party-ID header</synopsis>
315 <configOption name="timers_min_se" default="90">
316 <synopsis>Minimum session timers expiration period</synopsis>
318 Minimium session timer expiration period. Time in seconds.
319 </para></description>
321 <configOption name="timers" default="yes">
322 <synopsis>Session timers for SIP packets</synopsis>
325 <enum name="forced" />
327 <enum name="required" />
332 <configOption name="timers_sess_expires" default="1800">
333 <synopsis>Maximum session timer expiration period</synopsis>
335 Maximium session timer expiration period. Time in seconds.
336 </para></description>
338 <configOption name="transport">
339 <synopsis>Desired transport configuration</synopsis>
341 This will set the desired transport configuration to send SIP data through.
343 <warning><para>Not specifying a transport will <emphasis>DEFAULT</emphasis>
344 to the first configured transport in <filename>pjsip.conf</filename> which is
345 valid for the URI we are trying to contact.
347 <warning><para>Transport configuration is not affected by reloads. In order to
348 change transports, a full Asterisk restart is required</para></warning>
351 <configOption name="trust_id_inbound" default="no">
352 <synopsis>Accept identification information received from this endpoint</synopsis>
353 <description><para>This option determines whether Asterisk will accept
354 identification from the endpoint from headers such as P-Asserted-Identity
355 or Remote-Party-ID header. This option applies both to calls originating from the
356 endpoint and calls originating from Asterisk. If <literal>no</literal>, the
357 configured Caller-ID from pjsip.conf will always be used as the identity for
358 the endpoint.</para></description>
360 <configOption name="trust_id_outbound" default="no">
361 <synopsis>Send private identification details to the endpoint.</synopsis>
362 <description><para>This option determines whether res_pjsip will send private
363 identification information to the endpoint. If <literal>no</literal>,
364 private Caller-ID information will not be forwarded to the endpoint.
365 "Private" in this case refers to any method of restricting identification.
366 Example: setting <replaceable>callerid_privacy</replaceable> to any
367 <literal>prohib</literal> variation.
368 Example: If <replaceable>trust_id_inbound</replaceable> is set to
369 <literal>yes</literal>, the presence of a <literal>Privacy: id</literal>
370 header in a SIP request or response would indicate the identification
371 provided in the request is private.</para></description>
373 <configOption name="type">
374 <synopsis>Must be of type 'endpoint'.</synopsis>
376 <configOption name="use_ptime" default="no">
377 <synopsis>Use Endpoint's requested packetisation interval</synopsis>
379 <configOption name="use_avpf" default="no">
380 <synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this
383 If set to <literal>yes</literal>, res_pjsip will use use the AVPF or SAVPF RTP
384 profile for all media offers on outbound calls and media updates and will
385 decline media offers not using the AVPF or SAVPF profile.
387 If set to <literal>no</literal>, res_pjsip will use use the AVP or SAVP RTP
388 profile for all media offers on outbound calls and media updates and will
389 decline media offers not using the AVP or SAVP profile.
390 </para></description>
392 <configOption name="media_encryption" default="no">
393 <synopsis>Determines whether res_pjsip will use and enforce usage of media encryption
394 for this endpoint.</synopsis>
397 <enum name="no"><para>
398 res_pjsip will offer no encryption and allow no encryption to be setup.
400 <enum name="sdes"><para>
401 res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP
402 transport should be used in conjunction with this option to prevent
403 exposure of media encryption keys.
405 <enum name="dtls"><para>
406 res_pjsip will offer DTLS-SRTP setup.
411 <configOption name="inband_progress" default="no">
412 <synopsis>Determines whether chan_pjsip will indicate ringing using inband
415 If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
416 when told to indicate ringing and will immediately start sending ringing
419 If set to <literal>no</literal>, chan_pjsip will send a 180 Ringing when told
420 to indicate ringing and will NOT send it as audio.
421 </para></description>
423 <configOption name="call_group">
424 <synopsis>The numeric pickup groups for a channel.</synopsis>
426 Can be set to a comma separated list of numbers or ranges between the values
427 of 0-63 (maximum of 64 groups).
428 </para></description>
430 <configOption name="pickup_group">
431 <synopsis>The numeric pickup groups that a channel can pickup.</synopsis>
433 Can be set to a comma separated list of numbers or ranges between the values
434 of 0-63 (maximum of 64 groups).
435 </para></description>
437 <configOption name="named_call_group">
438 <synopsis>The named pickup groups for a channel.</synopsis>
440 Can be set to a comma separated list of case sensitive strings limited by
441 supported line length.
442 </para></description>
444 <configOption name="named_pickup_group">
445 <synopsis>The named pickup groups that a channel can pickup.</synopsis>
447 Can be set to a comma separated list of case sensitive strings limited by
448 supported line length.
449 </para></description>
451 <configOption name="device_state_busy_at" default="0">
452 <synopsis>The number of in-use channels which will cause busy to be returned as device state</synopsis>
454 When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the
455 PJSIP channel driver will return busy as the device state instead of in use.
456 </para></description>
458 <configOption name="t38_udptl" default="no">
459 <synopsis>Whether T.38 UDPTL support is enabled or not</synopsis>
461 If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted
463 </para></description>
465 <configOption name="t38_udptl_ec" default="none">
466 <synopsis>T.38 UDPTL error correction method</synopsis>
469 <enum name="none"><para>
470 No error correction should be used.
472 <enum name="fec"><para>
473 Forward error correction should be used.
475 <enum name="redundancy"><para>
476 Redundacy error correction should be used.
481 <configOption name="t38_udptl_maxdatagram" default="0">
482 <synopsis>T.38 UDPTL maximum datagram size</synopsis>
484 This option can be set to override the maximum datagram of a remote endpoint for broken
486 </para></description>
488 <configOption name="fax_detect" default="no">
489 <synopsis>Whether CNG tone detection is enabled</synopsis>
491 This option can be set to send the session to the fax extension when a CNG tone is
493 </para></description>
495 <configOption name="t38_udptl_nat" default="no">
496 <synopsis>Whether NAT support is enabled on UDPTL sessions</synopsis>
498 When enabled the UDPTL stack will send UDPTL packets to the source address of
500 </para></description>
502 <configOption name="t38_udptl_ipv6" default="no">
503 <synopsis>Whether IPv6 is used for UDPTL Sessions</synopsis>
505 When enabled the UDPTL stack will use IPv6.
506 </para></description>
508 <configOption name="tone_zone">
509 <synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis>
511 <configOption name="language">
512 <synopsis>Set the default language to use for channels created for this endpoint.</synopsis>
514 <configOption name="one_touch_recording" default="no">
515 <synopsis>Determines whether one-touch recording is allowed for this endpoint.</synopsis>
517 <ref type="configOption">recordonfeature</ref>
518 <ref type="configOption">recordofffeature</ref>
521 <configOption name="record_on_feature" default="automixmon">
522 <synopsis>The feature to enact when one-touch recording is turned on.</synopsis>
524 <para>When an INFO request for one-touch recording arrives with a Record header set to "on", this
525 feature will be enabled for the channel. The feature designated here can be any built-in
526 or dynamic feature defined in features.conf.</para>
527 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
530 <ref type="configOption">one_touch_recording</ref>
531 <ref type="configOption">recordofffeature</ref>
534 <configOption name="record_off_feature" default="automixmon">
535 <synopsis>The feature to enact when one-touch recording is turned off.</synopsis>
537 <para>When an INFO request for one-touch recording arrives with a Record header set to "off", this
538 feature will be enabled for the channel. The feature designated here can be any built-in
539 or dynamic feature defined in features.conf.</para>
540 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
543 <ref type="configOption">one_touch_recording</ref>
544 <ref type="configOption">recordonfeature</ref>
547 <configOption name="rtp_engine" default="asterisk">
548 <synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis>
550 <configOption name="allow_transfer" default="yes">
551 <synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis>
553 <configOption name="sdp_owner" default="-">
554 <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
556 <configOption name="sdp_session" default="Asterisk">
557 <synopsis>String used for the SDP session (s=) line.</synopsis>
559 <configOption name="tos_audio">
560 <synopsis>DSCP TOS bits for audio streams</synopsis>
562 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
563 </para></description>
565 <configOption name="tos_video">
566 <synopsis>DSCP TOS bits for video streams</synopsis>
568 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
569 </para></description>
571 <configOption name="cos_audio">
572 <synopsis>Priority for audio streams</synopsis>
574 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
575 </para></description>
577 <configOption name="cos_video">
578 <synopsis>Priority for video streams</synopsis>
580 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
581 </para></description>
583 <configOption name="allow_subscribe" default="yes">
584 <synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis>
586 <configOption name="sub_min_expiry" default="60">
587 <synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis>
589 <configOption name="from_user">
590 <synopsis>Username to use in From header for requests to this endpoint.</synopsis>
592 <configOption name="mwi_from_user">
593 <synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis>
595 <configOption name="from_domain">
596 <synopsis>Domain to user in From header for requests to this endpoint.</synopsis>
598 <configOption name="dtls_verify">
599 <synopsis>Verify that the provided peer certificate is valid</synopsis>
601 This option only applies if <replaceable>media_encryption</replaceable> is
602 set to <literal>dtls</literal>.
603 </para></description>
605 <configOption name="dtls_rekey">
606 <synopsis>Interval at which to renegotiate the TLS session and rekey the SRTP session</synopsis>
608 This option only applies if <replaceable>media_encryption</replaceable> is
609 set to <literal>dtls</literal>.
611 If this is not set or the value provided is 0 rekeying will be disabled.
612 </para></description>
614 <configOption name="dtls_cert_file">
615 <synopsis>Path to certificate file to present to peer</synopsis>
617 This option only applies if <replaceable>media_encryption</replaceable> is
618 set to <literal>dtls</literal>.
619 </para></description>
621 <configOption name="dtls_private_key">
622 <synopsis>Path to private key for certificate file</synopsis>
624 This option only applies if <replaceable>media_encryption</replaceable> is
625 set to <literal>dtls</literal>.
626 </para></description>
628 <configOption name="dtls_cipher">
629 <synopsis>Cipher to use for DTLS negotiation</synopsis>
631 This option only applies if <replaceable>media_encryption</replaceable> is
632 set to <literal>dtls</literal>.
634 Many options for acceptable ciphers. See link for more:
635 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
636 </para></description>
638 <configOption name="dtls_ca_file">
639 <synopsis>Path to certificate authority certificate</synopsis>
641 This option only applies if <replaceable>media_encryption</replaceable> is
642 set to <literal>dtls</literal>.
643 </para></description>
645 <configOption name="dtls_ca_path">
646 <synopsis>Path to a directory containing certificate authority certificates</synopsis>
648 This option only applies if <replaceable>media_encryption</replaceable> is
649 set to <literal>dtls</literal>.
650 </para></description>
652 <configOption name="dtls_setup">
653 <synopsis>Whether we are willing to accept connections, connect to the other party, or both.</synopsis>
656 This option only applies if <replaceable>media_encryption</replaceable> is
657 set to <literal>dtls</literal>.
660 <enum name="active"><para>
661 res_pjsip will make a connection to the peer.
663 <enum name="passive"><para>
664 res_pjsip will accept connections from the peer.
666 <enum name="actpass"><para>
667 res_pjsip will offer and accept connections from the peer.
672 <configOption name="srtp_tag_32">
673 <synopsis>Determines whether 32 byte tags should be used instead of 80 byte tags.</synopsis>
675 This option only applies if <replaceable>media_encryption</replaceable> is
676 set to <literal>sdes</literal> or <literal>dtls</literal>.
677 </para></description>
679 <configOption name="set_var">
680 <synopsis>Variable set on a channel involving the endpoint.</synopsis>
682 When a new channel is created using the endpoint set the specified
683 variable(s) on that channel. For multiple channel variables specify
684 multiple 'set_var'(s).
685 </para></description>
688 <configObject name="auth">
689 <synopsis>Authentication type</synopsis>
691 Authentication objects hold the authentication information for use
692 by other objects such as <literal>endpoints</literal> or <literal>registrations</literal>.
693 This also allows for multiple objects to use a single auth object. See
694 the <literal>auth_type</literal> config option for password style choices.
695 </para></description>
696 <configOption name="auth_type" default="userpass">
697 <synopsis>Authentication type</synopsis>
699 This option specifies which of the password style config options should be read
700 when trying to authenticate an endpoint inbound request. If set to <literal>userpass</literal>
701 then we'll read from the 'password' option. For <literal>md5</literal> we'll read
706 <enum name="userpass"/>
710 <configOption name="nonce_lifetime" default="32">
711 <synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
713 <configOption name="md5_cred">
714 <synopsis>MD5 Hash used for authentication.</synopsis>
715 <description><para>Only used when auth_type is <literal>md5</literal>.</para></description>
717 <configOption name="password">
718 <synopsis>PlainText password used for authentication.</synopsis>
719 <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
721 <configOption name="realm" default="asterisk">
722 <synopsis>SIP realm for endpoint</synopsis>
724 <configOption name="type">
725 <synopsis>Must be 'auth'</synopsis>
727 <configOption name="username">
728 <synopsis>Username to use for account</synopsis>
731 <configObject name="domain_alias">
732 <synopsis>Domain Alias</synopsis>
734 Signifies that a domain is an alias. If the domain on a session is
735 not found to match an AoR then this object is used to see if we have
736 an alias for the AoR to which the endpoint is binding. This objects
737 name as defined in configuration should be the domain alias and a
738 config option is provided to specify the domain to be aliased.
739 </para></description>
740 <configOption name="type">
741 <synopsis>Must be of type 'domain_alias'.</synopsis>
743 <configOption name="domain">
744 <synopsis>Domain to be aliased</synopsis>
747 <configObject name="transport">
748 <synopsis>SIP Transport</synopsis>
750 <emphasis>Transports</emphasis>
752 <para>There are different transports and protocol derivatives
753 supported by <literal>res_pjsip</literal>. They are in order of
754 preference: UDP, TCP, and WebSocket (WS).</para>
755 <note><para>Changes to transport configuration in pjsip.conf will only be
756 effected on a complete restart of Asterisk. A module reload
757 will not suffice.</para></note>
759 <configOption name="async_operations" default="1">
760 <synopsis>Number of simultaneous Asynchronous Operations</synopsis>
762 <configOption name="bind">
763 <synopsis>IP Address and optional port to bind to for this transport</synopsis>
765 <configOption name="ca_list_file">
766 <synopsis>File containing a list of certificates to read (TLS ONLY)</synopsis>
768 <configOption name="cert_file">
769 <synopsis>Certificate file for endpoint (TLS ONLY)</synopsis>
771 <configOption name="cipher">
772 <synopsis>Preferred Cryptography Cipher (TLS ONLY)</synopsis>
774 Many options for acceptable ciphers see link for more:
775 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
776 </para></description>
778 <configOption name="domain">
779 <synopsis>Domain the transport comes from</synopsis>
781 <configOption name="external_media_address">
782 <synopsis>External IP address to use in RTP handling</synopsis>
784 When a request or response is sent out, if the destination of the
785 message is outside the IP network defined in the option <literal>localnet</literal>,
786 and the media address in the SDP is within the localnet network, then the
787 media address in the SDP will be rewritten to the value defined for
788 <literal>external_media_address</literal>.
789 </para></description>
791 <configOption name="external_signaling_address">
792 <synopsis>External address for SIP signalling</synopsis>
794 <configOption name="external_signaling_port" default="0">
795 <synopsis>External port for SIP signalling</synopsis>
797 <configOption name="method">
798 <synopsis>Method of SSL transport (TLS ONLY)</synopsis>
801 <enum name="default" />
802 <enum name="unspecified" />
803 <enum name="tlsv1" />
804 <enum name="sslv2" />
805 <enum name="sslv3" />
806 <enum name="sslv23" />
810 <configOption name="local_net">
811 <synopsis>Network to consider local (used for NAT purposes).</synopsis>
812 <description><para>This must be in CIDR or dotted decimal format with the IP
813 and mask separated with a slash ('/').</para></description>
815 <configOption name="password">
816 <synopsis>Password required for transport</synopsis>
818 <configOption name="priv_key_file">
819 <synopsis>Private key file (TLS ONLY)</synopsis>
821 <configOption name="protocol" default="udp">
822 <synopsis>Protocol to use for SIP traffic</synopsis>
833 <configOption name="require_client_cert" default="false">
834 <synopsis>Require client certificate (TLS ONLY)</synopsis>
836 <configOption name="type">
837 <synopsis>Must be of type 'transport'.</synopsis>
839 <configOption name="verify_client" default="false">
840 <synopsis>Require verification of client certificate (TLS ONLY)</synopsis>
842 <configOption name="verify_server" default="false">
843 <synopsis>Require verification of server certificate (TLS ONLY)</synopsis>
845 <configOption name="tos" default="false">
846 <synopsis>Enable TOS for the signalling sent over this transport</synopsis>
848 <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
849 for more information on this parameter.</para>
850 <note><para>This option does not apply to the <replaceable>ws</replaceable>
851 or the <replaceable>wss</replaceable> protocols.</para></note>
854 <configOption name="cos" default="false">
855 <synopsis>Enable COS for the signalling sent over this transport</synopsis>
857 <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
858 for more information on this parameter.</para>
859 <note><para>This option does not apply to the <replaceable>ws</replaceable>
860 or the <replaceable>wss</replaceable> protocols.</para></note>
864 <configObject name="contact">
865 <synopsis>A way of creating an aliased name to a SIP URI</synopsis>
867 Contacts are a way to hide SIP URIs from the dialplan directly.
868 They are also used to make a group of contactable parties when
869 in use with <literal>AoR</literal> lists.
870 </para></description>
871 <configOption name="type">
872 <synopsis>Must be of type 'contact'.</synopsis>
874 <configOption name="uri">
875 <synopsis>SIP URI to contact peer</synopsis>
877 <configOption name="expiration_time">
878 <synopsis>Time to keep alive a contact</synopsis>
880 Time to keep alive a contact. String style specification.
881 </para></description>
883 <configOption name="qualify_frequency" default="0">
884 <synopsis>Interval at which to qualify a contact</synopsis>
886 Interval between attempts to qualify the contact for reachability.
887 If <literal>0</literal> never qualify. Time in seconds.
888 </para></description>
890 <configOption name="outbound_proxy">
891 <synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
893 If set the provided URI will be used as the outbound proxy when an
894 OPTIONS request is sent to a contact for qualify purposes.
895 </para></description>
897 <configOption name="path">
898 <synopsis>Stored Path vector for use in Route headers on outgoing requests.</synopsis>
901 <configObject name="aor">
902 <synopsis>The configuration for a location of an endpoint</synopsis>
904 An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
905 AoRs are specified, an endpoint will not be reachable by Asterisk.
906 Beyond that, an AoR has other uses within Asterisk, such as inbound
909 An <literal>AoR</literal> is a way to allow dialing a group
910 of <literal>Contacts</literal> that all use the same
911 <literal>endpoint</literal> for calls.
913 This can be used as another way of grouping a list of contacts to dial
914 rather than specifing them each directly when dialing via the dialplan.
915 This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
917 Registrations: For Asterisk to match an inbound registration to an endpoint,
918 the AoR object name must match the user portion of the SIP URI in the "To:"
919 header of the inbound SIP registration. That will usually be equivalent
920 to the "user name" set in your hard or soft phones configuration.
921 </para></description>
922 <configOption name="contact">
923 <synopsis>Permanent contacts assigned to AoR</synopsis>
925 Contacts specified will be called whenever referenced
926 by <literal>chan_pjsip</literal>.
928 Use a separate "contact=" entry for each contact required. Contacts
929 are specified using a SIP URI.
930 </para></description>
932 <configOption name="default_expiration" default="3600">
933 <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
935 <configOption name="mailboxes">
936 <synopsis>Allow subscriptions for the specified mailbox(es)</synopsis>
937 <description><para>This option applies when an external entity subscribes to an AoR
938 for Message Waiting Indications. The mailboxes specified will be subscribed to.
939 More than one mailbox can be specified with a comma-delimited string.
940 app_voicemail mailboxes must be specified as mailbox@context;
941 for example: mailboxes=6001@default. For mailboxes provided by external sources,
942 such as through the res_external_mwi module, you must specify strings supported by
945 For endpoints that cannot SUBSCRIBE for MWI, you can set the <literal>mailboxes</literal> option in your
946 endpoint configuration section to enable unsolicited MWI NOTIFYs to the endpoint.
947 </para></description>
949 <configOption name="maximum_expiration" default="7200">
950 <synopsis>Maximum time to keep an AoR</synopsis>
952 Maximium time to keep a peer with explicit expiration. Time in seconds.
953 </para></description>
955 <configOption name="max_contacts" default="0">
956 <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
958 Maximum number of contacts that can associate with this AoR. This value does
959 not affect the number of contacts that can be added with the "contact" option.
960 It only limits contacts added through external interaction, such as
963 <note><para>This should be set to <literal>1</literal> and
964 <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
965 wish to stick with the older <literal>chan_sip</literal> behaviour.
969 <configOption name="minimum_expiration" default="60">
970 <synopsis>Minimum keep alive time for an AoR</synopsis>
972 Minimum time to keep a peer with an explict expiration. Time in seconds.
973 </para></description>
975 <configOption name="remove_existing" default="no">
976 <synopsis>Determines whether new contacts replace existing ones.</synopsis>
978 On receiving a new registration to the AoR should it remove
979 the existing contact that was registered against it?
981 <note><para>This should be set to <literal>yes</literal> and
982 <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
983 wish to stick with the older <literal>chan_sip</literal> behaviour.
987 <configOption name="type">
988 <synopsis>Must be of type 'aor'.</synopsis>
990 <configOption name="qualify_frequency" default="0">
991 <synopsis>Interval at which to qualify an AoR</synopsis>
993 Interval between attempts to qualify the AoR for reachability.
994 If <literal>0</literal> never qualify. Time in seconds.
995 </para></description>
997 <configOption name="authenticate_qualify" default="no">
998 <synopsis>Authenticates a qualify request if needed</synopsis>
1000 If true and a qualify request receives a challenge or authenticate response
1001 authentication is attempted before declaring the contact available.
1002 </para></description>
1004 <configOption name="outbound_proxy">
1005 <synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
1007 If set the provided URI will be used as the outbound proxy when an
1008 OPTIONS request is sent to a contact for qualify purposes.
1009 </para></description>
1011 <configOption name="support_path">
1012 <synopsis>Enables Path support for REGISTER requests and Route support for other requests.</synopsis>
1014 When this option is enabled, the Path headers in register requests will be saved
1015 and its contents will be used in Route headers for outbound out-of-dialog requests
1016 and in Path headers for outbound 200 responses. Path support will also be indicated
1017 in the Supported header.
1018 </para></description>
1021 <configObject name="system">
1022 <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
1024 The settings in this section are global. In addition to being global, the values will
1025 not be re-evaluated when a reload is performed. This is because the values must be set
1026 before the SIP stack is initialized. The only way to reset these values is to either
1027 restart Asterisk, or unload res_pjsip.so and then load it again.
1028 </para></description>
1029 <configOption name="timer_t1" default="500">
1030 <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
1032 Timer T1 is the base for determining how long to wait before retransmitting
1033 requests that receive no response when using an unreliable transport (e.g. UDP).
1034 For more information on this timer, see RFC 3261, Section 17.1.1.1.
1035 </para></description>
1037 <configOption name="timer_b" default="32000">
1038 <synopsis>Set transaction timer B value (milliseconds).</synopsis>
1040 Timer B determines the maximum amount of time to wait after sending an INVITE
1041 request before terminating the transaction. It is recommended that this be set
1042 to 64 * Timer T1, but it may be set higher if desired. For more information on
1043 this timer, see RFC 3261, Section 17.1.1.1.
1044 </para></description>
1046 <configOption name="compact_headers" default="no">
1047 <synopsis>Use the short forms of common SIP header names.</synopsis>
1049 <configOption name="threadpool_initial_size" default="0">
1050 <synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
1052 <configOption name="threadpool_auto_increment" default="5">
1053 <synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
1055 <configOption name="threadpool_idle_timeout" default="60">
1056 <synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
1058 <configOption name="threadpool_max_size" default="0">
1059 <synopsis>Maximum number of threads in the res_pjsip threadpool.
1060 A value of 0 indicates no maximum.</synopsis>
1062 <configOption name="type">
1063 <synopsis>Must be of type 'system'.</synopsis>
1066 <configObject name="global">
1067 <synopsis>Options that apply globally to all SIP communications</synopsis>
1069 The settings in this section are global. Unlike options in the <literal>system</literal>
1070 section, these options can be refreshed by performing a reload.
1071 </para></description>
1072 <configOption name="max_forwards" default="70">
1073 <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
1075 <configOption name="type">
1076 <synopsis>Must be of type 'global'.</synopsis>
1078 <configOption name="user_agent" default="Asterisk <Asterisk Version>">
1079 <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
1081 <configOption name="default_outbound_endpoint" default="default_outbound_endpoint">
1082 <synopsis>Endpoint to use when sending an outbound request to a URI without a specified endpoint.</synopsis>
1088 <manager name="PJSIPQualify" language="en_US">
1090 Qualify a chan_pjsip endpoint.
1093 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1094 <parameter name="Endpoint" required="true">
1095 <para>The endpoint you want to qualify.</para>
1099 <para>Qualify a chan_pjsip endpoint.</para>
1102 <manager name="PJSIPShowEndpoints" language="en_US">
1104 Lists PJSIP endpoints.
1109 Provides a listing of all endpoints. For each endpoint an <literal>EndpointList</literal> event
1110 is raised that contains relevant attributes and status information. Once all
1111 endpoints have been listed an <literal>EndpointListComplete</literal> event is issued.
1115 <manager name="PJSIPShowEndpoint" language="en_US">
1117 Detail listing of an endpoint and its objects.
1120 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1121 <parameter name="Endpoint" required="true">
1122 <para>The endpoint to list.</para>
1127 Provides a detailed listing of options for a given endpoint. Events are issued
1128 showing the configuration and status of the endpoint and associated objects. These
1129 events include <literal>EndpointDetail</literal>, <literal>AorDetail</literal>,
1130 <literal>AuthDetail</literal>, <literal>TransportDetail</literal>, and
1131 <literal>IdentifyDetail</literal>. Some events may be listed multiple times if multiple objects are
1132 associated (for instance AoRs). Once all detail events have been raised a final
1133 <literal>EndpointDetailComplete</literal> event is issued.
1139 #define MOD_DATA_CONTACT "contact"
1141 static pjsip_endpoint *ast_pjsip_endpoint;
1143 static struct ast_threadpool *sip_threadpool;
1145 static int register_service(void *data)
1147 pjsip_module **module = data;
1148 if (!ast_pjsip_endpoint) {
1149 ast_log(LOG_ERROR, "There is no PJSIP endpoint. Unable to register services\n");
1152 if (pjsip_endpt_register_module(ast_pjsip_endpoint, *module) != PJ_SUCCESS) {
1153 ast_log(LOG_ERROR, "Unable to register module %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1156 ast_debug(1, "Registered SIP service %.*s (%p)\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name), *module);
1157 ast_module_ref(ast_module_info->self);
1161 int ast_sip_register_service(pjsip_module *module)
1163 return ast_sip_push_task_synchronous(NULL, register_service, &module);
1166 static int unregister_service(void *data)
1168 pjsip_module **module = data;
1169 ast_module_unref(ast_module_info->self);
1170 if (!ast_pjsip_endpoint) {
1173 pjsip_endpt_unregister_module(ast_pjsip_endpoint, *module);
1174 ast_debug(1, "Unregistered SIP service %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1178 void ast_sip_unregister_service(pjsip_module *module)
1180 ast_sip_push_task_synchronous(NULL, unregister_service, &module);
1183 static struct ast_sip_authenticator *registered_authenticator;
1185 int ast_sip_register_authenticator(struct ast_sip_authenticator *auth)
1187 if (registered_authenticator) {
1188 ast_log(LOG_WARNING, "Authenticator %p is already registered. Cannot register a new one\n", registered_authenticator);
1191 registered_authenticator = auth;
1192 ast_debug(1, "Registered SIP authenticator module %p\n", auth);
1193 ast_module_ref(ast_module_info->self);
1197 void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth)
1199 if (registered_authenticator != auth) {
1200 ast_log(LOG_WARNING, "Trying to unregister authenticator %p but authenticator %p registered\n",
1201 auth, registered_authenticator);
1204 registered_authenticator = NULL;
1205 ast_debug(1, "Unregistered SIP authenticator %p\n", auth);
1206 ast_module_unref(ast_module_info->self);
1209 int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1211 if (!registered_authenticator) {
1212 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is not required\n");
1216 return registered_authenticator->requires_authentication(endpoint, rdata);
1219 enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
1220 pjsip_rx_data *rdata, pjsip_tx_data *tdata)
1222 if (!registered_authenticator) {
1223 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is successful\n");
1226 return registered_authenticator->check_authentication(endpoint, rdata, tdata);
1229 static struct ast_sip_outbound_authenticator *registered_outbound_authenticator;
1231 int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1233 if (registered_outbound_authenticator) {
1234 ast_log(LOG_WARNING, "Outbound authenticator %p is already registered. Cannot register a new one\n", registered_outbound_authenticator);
1237 registered_outbound_authenticator = auth;
1238 ast_debug(1, "Registered SIP outbound authenticator module %p\n", auth);
1239 ast_module_ref(ast_module_info->self);
1243 void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1245 if (registered_outbound_authenticator != auth) {
1246 ast_log(LOG_WARNING, "Trying to unregister outbound authenticator %p but outbound authenticator %p registered\n",
1247 auth, registered_outbound_authenticator);
1250 registered_outbound_authenticator = NULL;
1251 ast_debug(1, "Unregistered SIP outbound authenticator %p\n", auth);
1252 ast_module_unref(ast_module_info->self);
1255 int ast_sip_create_request_with_auth(const struct ast_sip_auth_vector *auths, pjsip_rx_data *challenge,
1256 pjsip_transaction *tsx, pjsip_tx_data **new_request)
1258 if (!registered_outbound_authenticator) {
1259 ast_log(LOG_WARNING, "No SIP outbound authenticator registered. Cannot respond to authentication challenge\n");
1262 return registered_outbound_authenticator->create_request_with_auth(auths, challenge, tsx, new_request);
1265 struct endpoint_identifier_list {
1266 struct ast_sip_endpoint_identifier *identifier;
1267 AST_RWLIST_ENTRY(endpoint_identifier_list) list;
1270 static AST_RWLIST_HEAD_STATIC(endpoint_identifiers, endpoint_identifier_list);
1272 int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1274 struct endpoint_identifier_list *id_list_item;
1275 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1277 id_list_item = ast_calloc(1, sizeof(*id_list_item));
1278 if (!id_list_item) {
1279 ast_log(LOG_ERROR, "Unabled to add endpoint identifier. Out of memory.\n");
1282 id_list_item->identifier = identifier;
1284 AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list);
1285 ast_debug(1, "Registered endpoint identifier %p\n", identifier);
1287 ast_module_ref(ast_module_info->self);
1291 void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1293 struct endpoint_identifier_list *iter;
1294 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1295 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) {
1296 if (iter->identifier == identifier) {
1297 AST_RWLIST_REMOVE_CURRENT(list);
1299 ast_debug(1, "Unregistered endpoint identifier %p\n", identifier);
1300 ast_module_unref(ast_module_info->self);
1304 AST_RWLIST_TRAVERSE_SAFE_END;
1307 struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata)
1309 struct endpoint_identifier_list *iter;
1310 struct ast_sip_endpoint *endpoint = NULL;
1311 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1312 AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) {
1313 ast_assert(iter->identifier->identify_endpoint != NULL);
1314 endpoint = iter->identifier->identify_endpoint(rdata);
1322 AST_RWLIST_HEAD_STATIC(endpoint_formatters, ast_sip_endpoint_formatter);
1324 int ast_sip_register_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
1326 SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1327 AST_RWLIST_INSERT_TAIL(&endpoint_formatters, obj, next);
1328 ast_module_ref(ast_module_info->self);
1332 void ast_sip_unregister_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
1334 struct ast_sip_endpoint_formatter *i;
1335 SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1336 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_formatters, i, next) {
1338 AST_RWLIST_REMOVE_CURRENT(next);
1339 ast_module_unref(ast_module_info->self);
1343 AST_RWLIST_TRAVERSE_SAFE_END;
1346 int ast_sip_format_endpoint_ami(struct ast_sip_endpoint *endpoint,
1347 struct ast_sip_ami *ami, int *count)
1350 struct ast_sip_endpoint_formatter *i;
1351 SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1353 AST_RWLIST_TRAVERSE(&endpoint_formatters, i, next) {
1354 if (i->format_ami && ((res = i->format_ami(endpoint, ami)) < 0)) {
1365 pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
1367 return ast_pjsip_endpoint;
1370 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)
1372 pj_str_t tmp, local_addr;
1374 pjsip_sip_uri *sip_uri;
1375 pjsip_transport_type_e type = PJSIP_TRANSPORT_UNSPECIFIED;
1377 char uuid_str[AST_UUID_STR_LEN];
1379 if (ast_strlen_zero(user)) {
1380 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1384 user = ast_uuid_to_str(uuid, uuid_str, sizeof(uuid_str));
1387 /* Parse the provided target URI so we can determine what transport it will end up using */
1388 pj_strdup_with_null(pool, &tmp, target);
1390 if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) ||
1391 (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
1395 sip_uri = pjsip_uri_get_uri(uri);
1397 /* Determine the transport type to use */
1398 if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
1399 type = PJSIP_TRANSPORT_TLS;
1400 } else if (!sip_uri->transport_param.slen) {
1401 type = PJSIP_TRANSPORT_UDP;
1403 type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
1406 if (type == PJSIP_TRANSPORT_UNSPECIFIED) {
1410 /* If the host is IPv6 turn the transport into an IPv6 version */
1411 if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1412 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1415 if (!ast_strlen_zero(domain)) {
1416 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1417 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1421 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1422 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1426 /* Get the local bound address for the transport that will be used when communicating with the provided URI */
1427 if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), pool, type, selector,
1428 &local_addr, &local_port) != PJ_SUCCESS) {
1430 /* If no local address can be retrieved using the transport manager use the host one */
1431 pj_strdup(pool, &local_addr, pj_gethostname());
1432 local_port = pjsip_transport_get_default_port_for_type(PJSIP_TRANSPORT_UDP);
1435 /* If IPv6 was specified in the transport, set the proper type */
1436 if (pj_strchr(&local_addr, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1437 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1440 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1441 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1442 "<sip:%s@%s%.*s%s:%d%s%s>",
1444 (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1445 (int)local_addr.slen,
1447 (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1449 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1450 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1455 static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
1457 RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
1458 const char *transport_name = endpoint->transport;
1460 if (ast_strlen_zero(transport_name)) {
1464 transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name);
1466 if (!transport || !transport->state) {
1467 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport '%s' for endpoint '%s'\n",
1468 transport_name, ast_sorcery_object_get_id(endpoint));
1472 if (transport->state->transport) {
1473 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1474 selector->u.transport = transport->state->transport;
1475 } else if (transport->state->factory) {
1476 selector->type = PJSIP_TPSELECTOR_LISTENER;
1477 selector->u.listener = transport->state->factory;
1485 pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
1487 char enclosed_uri[PJSIP_MAX_URL_SIZE];
1488 pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri, target_uri;
1489 pjsip_dialog *dlg = NULL;
1490 const char *outbound_proxy = endpoint->outbound_proxy;
1491 pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1492 static const pj_str_t HCONTACT = { "Contact", 7 };
1494 snprintf(enclosed_uri, sizeof(enclosed_uri), "<%s>", uri);
1495 pj_cstr(&remote_uri, enclosed_uri);
1497 pj_cstr(&target_uri, uri);
1499 if (pjsip_dlg_create_uac(pjsip_ua_instance(), &local_uri, NULL, &remote_uri, &target_uri, &dlg) != PJ_SUCCESS) {
1503 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1504 pjsip_dlg_terminate(dlg);
1508 if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
1509 pjsip_dlg_terminate(dlg);
1513 /* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
1514 pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
1515 dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
1516 dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
1518 /* If a request user has been specified and we are permitted to change it, do so */
1519 if (!ast_strlen_zero(request_user)) {
1520 pjsip_sip_uri *sip_uri;
1522 if (PJSIP_URI_SCHEME_IS_SIP(dlg->target) || PJSIP_URI_SCHEME_IS_SIPS(dlg->target)) {
1523 sip_uri = pjsip_uri_get_uri(dlg->target);
1524 pj_strdup2(dlg->pool, &sip_uri->user, request_user);
1526 if (PJSIP_URI_SCHEME_IS_SIP(dlg->remote.info->uri) || PJSIP_URI_SCHEME_IS_SIPS(dlg->remote.info->uri)) {
1527 sip_uri = pjsip_uri_get_uri(dlg->remote.info->uri);
1528 pj_strdup2(dlg->pool, &sip_uri->user, request_user);
1532 /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
1535 pjsip_dlg_set_transport(dlg, &selector);
1537 if (!ast_strlen_zero(outbound_proxy)) {
1538 pjsip_route_hdr route_set, *route;
1539 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1542 pj_list_init(&route_set);
1544 pj_strdup2_with_null(dlg->pool, &tmp, outbound_proxy);
1545 if (!(route = pjsip_parse_hdr(dlg->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1547 pjsip_dlg_terminate(dlg);
1550 pj_list_insert_nodes_before(&route_set, route);
1552 pjsip_dlg_set_route_set(dlg, &route_set);
1560 pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1564 pjsip_transport_type_e type = rdata->tp_info.transport->key.type;
1567 contact.ptr = pj_pool_alloc(rdata->tp_info.pool, PJSIP_MAX_URL_SIZE);
1568 contact.slen = pj_ansi_snprintf(contact.ptr, PJSIP_MAX_URL_SIZE,
1569 "<sip:%s%.*s%s:%d%s%s>",
1570 (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1571 (int)rdata->tp_info.transport->local_name.host.slen,
1572 rdata->tp_info.transport->local_name.host.ptr,
1573 (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1574 rdata->tp_info.transport->local_name.port,
1575 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1576 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1578 status = pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, &contact, &dlg);
1579 if (status != PJ_SUCCESS) {
1580 char err[PJ_ERR_MSG_SIZE];
1582 pj_strerror(status, err, sizeof(err));
1583 ast_log(LOG_ERROR, "Could not create dialog with endpoint %s. %s\n",
1584 ast_sorcery_object_get_id(endpoint), err);
1591 /* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
1592 static const pjsip_method info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
1593 static const pjsip_method message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
1597 const pjsip_method *pmethod;
1599 { "INVITE", &pjsip_invite_method },
1600 { "CANCEL", &pjsip_cancel_method },
1601 { "ACK", &pjsip_ack_method },
1602 { "BYE", &pjsip_bye_method },
1603 { "REGISTER", &pjsip_register_method },
1604 { "OPTIONS", &pjsip_options_method },
1605 { "SUBSCRIBE", &pjsip_subscribe_method },
1606 { "NOTIFY", &pjsip_notify_method },
1607 { "PUBLISH", &pjsip_publish_method },
1608 { "INFO", &info_method },
1609 { "MESSAGE", &message_method },
1612 static const pjsip_method *get_pjsip_method(const char *method)
1615 for (i = 0; i < ARRAY_LEN(methods); ++i) {
1616 if (!strcmp(method, methods[i].method)) {
1617 return methods[i].pmethod;
1623 static int create_in_dialog_request(const pjsip_method *method, struct pjsip_dialog *dlg, pjsip_tx_data **tdata)
1625 if (pjsip_dlg_create_request(dlg, method, -1, tdata) != PJ_SUCCESS) {
1626 ast_log(LOG_WARNING, "Unable to create in-dialog request.\n");
1633 static pj_bool_t supplement_on_rx_request(pjsip_rx_data *rdata);
1634 static pjsip_module supplement_module = {
1635 .name = { "Out of dialog supplement hook", 29 },
1637 .priority = PJSIP_MOD_PRIORITY_APPLICATION - 1,
1638 .on_rx_request = supplement_on_rx_request,
1641 static int create_out_of_dialog_request(const pjsip_method *method, struct ast_sip_endpoint *endpoint,
1642 const char *uri, struct ast_sip_contact *provided_contact, pjsip_tx_data **tdata)
1644 RAII_VAR(struct ast_sip_contact *, contact, ao2_bump(provided_contact), ao2_cleanup);
1645 pj_str_t remote_uri;
1648 pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1650 if (ast_strlen_zero(uri)) {
1651 if (!endpoint && !contact) {
1652 ast_log(LOG_ERROR, "An endpoint and/or uri must be specified\n");
1657 contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
1659 if (!contact || ast_strlen_zero(contact->uri)) {
1660 ast_log(LOG_ERROR, "Unable to retrieve contact for endpoint %s\n",
1661 ast_sorcery_object_get_id(endpoint));
1665 pj_cstr(&remote_uri, contact->uri);
1667 pj_cstr(&remote_uri, uri);
1671 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1672 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
1673 ast_sorcery_object_get_id(endpoint));
1678 pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
1681 ast_log(LOG_ERROR, "Unable to create PJLIB memory pool\n");
1685 if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL,
1686 endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
1687 ast_log(LOG_ERROR, "Unable to create From header for %.*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 (pjsip_endpt_create_request(ast_sip_get_pjsip_endpoint(), method, &remote_uri,
1694 &from, &remote_uri, &from, NULL, -1, NULL, tdata) != PJ_SUCCESS) {
1695 ast_log(LOG_ERROR, "Unable to create outbound %.*s request 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 /* If an outbound proxy is specified on the endpoint apply it to this request */
1702 if (endpoint && !ast_strlen_zero(endpoint->outbound_proxy) &&
1703 ast_sip_set_outbound_proxy((*tdata), endpoint->outbound_proxy)) {
1704 ast_log(LOG_ERROR, "Unable to apply outbound proxy on request %.*s to endpoint %s\n",
1705 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1706 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1710 ast_sip_mod_data_set((*tdata)->pool, (*tdata)->mod_data, supplement_module.id, MOD_DATA_CONTACT, ao2_bump(contact));
1712 /* We can release this pool since request creation copied all the necessary
1713 * data into the outbound request's pool
1715 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1719 int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
1720 struct ast_sip_endpoint *endpoint, const char *uri,
1721 struct ast_sip_contact *contact, pjsip_tx_data **tdata)
1723 const pjsip_method *pmethod = get_pjsip_method(method);
1726 ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
1731 return create_in_dialog_request(pmethod, dlg, tdata);
1733 return create_out_of_dialog_request(pmethod, endpoint, uri, contact, tdata);
1737 AST_RWLIST_HEAD_STATIC(supplements, ast_sip_supplement);
1739 int ast_sip_register_supplement(struct ast_sip_supplement *supplement)
1741 struct ast_sip_supplement *iter;
1743 SCOPED_LOCK(lock, &supplements, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1745 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&supplements, iter, next) {
1746 if (iter->priority > supplement->priority) {
1747 AST_RWLIST_INSERT_BEFORE_CURRENT(supplement, next);
1752 AST_RWLIST_TRAVERSE_SAFE_END;
1755 AST_RWLIST_INSERT_TAIL(&supplements, supplement, next);
1757 ast_module_ref(ast_module_info->self);
1761 void ast_sip_unregister_supplement(struct ast_sip_supplement *supplement)
1763 struct ast_sip_supplement *iter;
1764 SCOPED_LOCK(lock, &supplements, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1765 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&supplements, iter, next) {
1766 if (supplement == iter) {
1767 AST_RWLIST_REMOVE_CURRENT(next);
1768 ast_module_unref(ast_module_info->self);
1772 AST_RWLIST_TRAVERSE_SAFE_END;
1775 static int send_in_dialog_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg)
1777 if (pjsip_dlg_send_request(dlg, tdata, -1, NULL) != PJ_SUCCESS) {
1778 ast_log(LOG_WARNING, "Unable to send in-dialog request.\n");
1784 static pj_bool_t does_method_match(const pj_str_t *message_method, const char *supplement_method)
1788 if (ast_strlen_zero(supplement_method)) {
1792 pj_cstr(&method, supplement_method);
1794 return pj_stristr(&method, message_method) ? PJ_TRUE : PJ_FALSE;
1797 /*! \brief Structure to hold information about an outbound request */
1798 struct send_request_data {
1799 struct ast_sip_endpoint *endpoint; /*! The endpoint associated with this request */
1800 void *token; /*! Information to be provided to the callback upon receipt of a response */
1801 void (*callback)(void *token, pjsip_event *e); /*! The callback to be called upon receipt of a response */
1804 static void send_request_data_destroy(void *obj)
1806 struct send_request_data *req_data = obj;
1807 ao2_cleanup(req_data->endpoint);
1810 static struct send_request_data *send_request_data_alloc(struct ast_sip_endpoint *endpoint,
1811 void *token, void (*callback)(void *token, pjsip_event *e))
1813 struct send_request_data *req_data = ao2_alloc(sizeof(*req_data), send_request_data_destroy);
1819 req_data->endpoint = ao2_bump(endpoint);
1820 req_data->token = token;
1821 req_data->callback = callback;
1826 static void send_request_cb(void *token, pjsip_event *e)
1828 RAII_VAR(struct send_request_data *, req_data, token, ao2_cleanup);
1829 pjsip_transaction *tsx = e->body.tsx_state.tsx;
1830 pjsip_rx_data *challenge = e->body.tsx_state.src.rdata;
1831 pjsip_tx_data *tdata;
1832 struct ast_sip_supplement *supplement;
1834 AST_RWLIST_RDLOCK(&supplements);
1835 AST_LIST_TRAVERSE(&supplements, supplement, next) {
1836 if (supplement->incoming_response && does_method_match(&challenge->msg_info.cseq->method.name, supplement->method)) {
1837 supplement->incoming_response(req_data->endpoint, challenge);
1840 AST_RWLIST_UNLOCK(&supplements);
1842 if (tsx->status_code == 401 || tsx->status_code == 407) {
1843 if (!ast_sip_create_request_with_auth(&req_data->endpoint->outbound_auths, challenge, tsx, &tdata)) {
1844 pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, req_data->token, req_data->callback);
1849 if (req_data->callback) {
1850 req_data->callback(req_data->token, e);
1854 static int send_out_of_dialog_request(pjsip_tx_data *tdata, struct ast_sip_endpoint *endpoint,
1855 void *token, void (*callback)(void *token, pjsip_event *e))
1857 struct ast_sip_supplement *supplement;
1858 struct send_request_data *req_data = send_request_data_alloc(endpoint, token, callback);
1859 struct ast_sip_contact *contact = ast_sip_mod_data_get(tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT);
1865 AST_RWLIST_RDLOCK(&supplements);
1866 AST_LIST_TRAVERSE(&supplements, supplement, next) {
1867 if (supplement->outgoing_request && does_method_match(&tdata->msg->line.req.method.name, supplement->method)) {
1868 supplement->outgoing_request(endpoint, contact, tdata);
1871 AST_RWLIST_UNLOCK(&supplements);
1873 ast_sip_mod_data_set(tdata->pool, tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT, NULL);
1874 ao2_cleanup(contact);
1876 if (pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, req_data, send_request_cb) != PJ_SUCCESS) {
1877 ast_log(LOG_ERROR, "Error attempting to send outbound %.*s request to endpoint %s\n",
1878 (int) pj_strlen(&tdata->msg->line.req.method.name),
1879 pj_strbuf(&tdata->msg->line.req.method.name),
1880 ast_sorcery_object_get_id(endpoint));
1881 ao2_cleanup(req_data);
1888 int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg,
1889 struct ast_sip_endpoint *endpoint, void *token,
1890 void (*callback)(void *token, pjsip_event *e))
1892 ast_assert(tdata->msg->type == PJSIP_REQUEST_MSG);
1895 return send_in_dialog_request(tdata, dlg);
1897 return send_out_of_dialog_request(tdata, endpoint, token, callback);
1901 int ast_sip_set_outbound_proxy(pjsip_tx_data *tdata, const char *proxy)
1903 pjsip_route_hdr *route;
1904 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1907 pj_strdup2_with_null(tdata->pool, &tmp, proxy);
1908 if (!(route = pjsip_parse_hdr(tdata->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1912 pj_list_insert_nodes_before(&tdata->msg->hdr, (pjsip_hdr*)route);
1917 int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value)
1921 pjsip_generic_string_hdr *hdr;
1923 pj_cstr(&hdr_name, name);
1924 pj_cstr(&hdr_value, value);
1926 hdr = pjsip_generic_string_hdr_create(tdata->pool, &hdr_name, &hdr_value);
1928 pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) hdr);
1932 static pjsip_msg_body *ast_body_to_pjsip_body(pj_pool_t *pool, const struct ast_sip_body *body)
1938 pj_cstr(&type, body->type);
1939 pj_cstr(&subtype, body->subtype);
1940 pj_cstr(&body_text, body->body_text);
1942 return pjsip_msg_body_create(pool, &type, &subtype, &body_text);
1945 int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body)
1947 pjsip_msg_body *pjsip_body = ast_body_to_pjsip_body(tdata->pool, body);
1948 tdata->msg->body = pjsip_body;
1952 int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies)
1955 /* NULL for type and subtype automatically creates "multipart/mixed" */
1956 pjsip_msg_body *body = pjsip_multipart_create(tdata->pool, NULL, NULL);
1958 for (i = 0; i < num_bodies; ++i) {
1959 pjsip_multipart_part *part = pjsip_multipart_create_part(tdata->pool);
1960 part->body = ast_body_to_pjsip_body(tdata->pool, bodies[i]);
1961 pjsip_multipart_add_part(tdata->pool, body, part);
1964 tdata->msg->body = body;
1968 int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text)
1970 size_t combined_size = strlen(body_text) + tdata->msg->body->len;
1971 struct ast_str *body_buffer = ast_str_alloca(combined_size);
1973 ast_str_set(&body_buffer, 0, "%.*s%s", (int) tdata->msg->body->len, (char *) tdata->msg->body->data, body_text);
1975 tdata->msg->body->data = pj_pool_alloc(tdata->pool, combined_size);
1976 pj_memcpy(tdata->msg->body->data, ast_str_buffer(body_buffer), combined_size);
1977 tdata->msg->body->len = combined_size;
1982 struct ast_taskprocessor *ast_sip_create_serializer(void)
1984 struct ast_taskprocessor *serializer;
1985 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1986 char name[AST_UUID_STR_LEN];
1992 ast_uuid_to_str(uuid, name, sizeof(name));
1994 serializer = ast_threadpool_serializer(name, sip_threadpool);
2001 int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
2004 return ast_taskprocessor_push(serializer, sip_task, task_data);
2006 return ast_threadpool_push(sip_threadpool, sip_task, task_data);
2010 struct sync_task_data {
2015 int (*task)(void *);
2019 static int sync_task(void *data)
2021 struct sync_task_data *std = data;
2022 std->fail = std->task(std->task_data);
2024 ast_mutex_lock(&std->lock);
2026 ast_cond_signal(&std->cond);
2027 ast_mutex_unlock(&std->lock);
2031 int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
2033 /* This method is an onion */
2034 struct sync_task_data std;
2036 if (ast_sip_thread_is_servant()) {
2037 return sip_task(task_data);
2040 ast_mutex_init(&std.lock);
2041 ast_cond_init(&std.cond, NULL);
2042 std.fail = std.complete = 0;
2043 std.task = sip_task;
2044 std.task_data = task_data;
2047 if (ast_taskprocessor_push(serializer, sync_task, &std)) {
2051 if (ast_threadpool_push(sip_threadpool, sync_task, &std)) {
2056 ast_mutex_lock(&std.lock);
2057 while (!std.complete) {
2058 ast_cond_wait(&std.cond, &std.lock);
2060 ast_mutex_unlock(&std.lock);
2062 ast_mutex_destroy(&std.lock);
2063 ast_cond_destroy(&std.cond);
2067 void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
2069 size_t chars_to_copy = MIN(size - 1, pj_strlen(src));
2070 memcpy(dest, pj_strbuf(src), chars_to_copy);
2071 dest[chars_to_copy] = '\0';
2074 int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype)
2076 pjsip_media_type compare;
2078 if (!content_type) {
2082 pjsip_media_type_init2(&compare, type, subtype);
2084 return pjsip_media_type_cmp(content_type, &compare, 0) ? 0 : -1;
2087 pj_caching_pool caching_pool;
2088 pj_pool_t *memory_pool;
2089 pj_thread_t *monitor_thread;
2090 static int monitor_continue;
2092 static void *monitor_thread_exec(void *endpt)
2094 while (monitor_continue) {
2095 const pj_time_val delay = {0, 10};
2096 pjsip_endpt_handle_events(ast_pjsip_endpoint, &delay);
2101 static void stop_monitor_thread(void)
2103 monitor_continue = 0;
2104 pj_thread_join(monitor_thread);
2107 AST_THREADSTORAGE(pj_thread_storage);
2108 AST_THREADSTORAGE(servant_id_storage);
2109 #define SIP_SERVANT_ID 0x5E2F1D
2111 static void sip_thread_start(void)
2113 pj_thread_desc *desc;
2114 pj_thread_t *thread;
2115 uint32_t *servant_id;
2117 servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
2119 ast_log(LOG_ERROR, "Could not set SIP servant ID in thread-local storage.\n");
2122 *servant_id = SIP_SERVANT_ID;
2124 desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc));
2126 ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n");
2129 pj_bzero(*desc, sizeof(*desc));
2131 if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) {
2132 ast_log(LOG_ERROR, "Couldn't register thread with PJLIB.\n");
2136 int ast_sip_thread_is_servant(void)
2138 uint32_t *servant_id;
2140 servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
2145 return *servant_id == SIP_SERVANT_ID;
2148 void *ast_sip_dict_get(void *ht, const char *key)
2150 unsigned int hval = 0;
2156 return pj_hash_get(ht, key, PJ_HASH_KEY_STRING, &hval);
2159 void *ast_sip_dict_set(pj_pool_t* pool, void *ht,
2160 const char *key, void *val)
2163 ht = pj_hash_create(pool, 11);
2166 pj_hash_set(pool, ht, key, PJ_HASH_KEY_STRING, 0, val);
2171 static pj_bool_t supplement_on_rx_request(pjsip_rx_data *rdata)
2173 struct ast_sip_supplement *supplement;
2175 if (pjsip_rdata_get_dlg(rdata)) {
2179 AST_RWLIST_RDLOCK(&supplements);
2180 AST_LIST_TRAVERSE(&supplements, supplement, next) {
2181 if (supplement->incoming_request && does_method_match(&rdata->msg_info.msg->line.req.method.name, supplement->method)) {
2182 supplement->incoming_request(ast_pjsip_rdata_get_endpoint(rdata), rdata);
2185 AST_RWLIST_UNLOCK(&supplements);
2190 int ast_sip_send_response(pjsip_response_addr *res_addr, pjsip_tx_data *tdata, struct ast_sip_endpoint *sip_endpoint)
2192 struct ast_sip_supplement *supplement;
2193 pjsip_cseq_hdr *cseq = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_CSEQ, NULL);
2194 struct ast_sip_contact *contact = ast_sip_mod_data_get(tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT);
2196 AST_RWLIST_RDLOCK(&supplements);
2197 AST_LIST_TRAVERSE(&supplements, supplement, next) {
2198 if (supplement->outgoing_response && does_method_match(&cseq->method.name, supplement->method)) {
2199 supplement->outgoing_response(sip_endpoint, contact, tdata);
2202 AST_RWLIST_UNLOCK(&supplements);
2204 ast_sip_mod_data_set(tdata->pool, tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT, NULL);
2205 ao2_cleanup(contact);
2207 return pjsip_endpt_send_response(ast_sip_get_pjsip_endpoint(), res_addr, tdata, NULL, NULL);
2210 int ast_sip_create_response(const pjsip_rx_data *rdata, int st_code,
2211 struct ast_sip_contact *contact, pjsip_tx_data **tdata)
2213 int res = pjsip_endpt_create_response(ast_sip_get_pjsip_endpoint(), rdata, st_code, NULL, tdata);
2216 ast_sip_mod_data_set((*tdata)->pool, (*tdata)->mod_data, supplement_module.id, MOD_DATA_CONTACT, ao2_bump(contact));
2222 static void remove_request_headers(pjsip_endpoint *endpt)
2224 const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt);
2225 pjsip_hdr *iter = request_headers->next;
2227 while (iter != request_headers) {
2228 pjsip_hdr *to_erase = iter;
2230 pj_list_erase(to_erase);
2234 static int load_module(void)
2236 /* The third parameter is just copied from
2237 * example code from PJLIB. This can be adjusted
2241 struct ast_threadpool_options options;
2243 if (pj_init() != PJ_SUCCESS) {
2244 return AST_MODULE_LOAD_DECLINE;
2247 if (pjlib_util_init() != PJ_SUCCESS) {
2249 return AST_MODULE_LOAD_DECLINE;
2252 pj_caching_pool_init(&caching_pool, NULL, 1024 * 1024);
2253 if (pjsip_endpt_create(&caching_pool.factory, "SIP", &ast_pjsip_endpoint) != PJ_SUCCESS) {
2254 ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n");
2255 pj_caching_pool_destroy(&caching_pool);
2256 return AST_MODULE_LOAD_DECLINE;
2259 /* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that,
2260 * we need to stop PJSIP from doing it automatically
2262 remove_request_headers(ast_pjsip_endpoint);
2264 memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL);
2266 ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n");
2267 pjsip_endpt_destroy(ast_pjsip_endpoint);
2268 ast_pjsip_endpoint = NULL;
2269 pj_caching_pool_destroy(&caching_pool);
2270 return AST_MODULE_LOAD_DECLINE;
2273 if (ast_sip_initialize_system()) {
2274 ast_log(LOG_ERROR, "Failed to initialize SIP 'system' configuration section. Aborting load\n");
2275 pj_pool_release(memory_pool);
2277 pjsip_endpt_destroy(ast_pjsip_endpoint);
2278 ast_pjsip_endpoint = NULL;
2279 pj_caching_pool_destroy(&caching_pool);
2280 return AST_MODULE_LOAD_DECLINE;
2283 sip_get_threadpool_options(&options);
2284 options.thread_start = sip_thread_start;
2285 sip_threadpool = ast_threadpool_create("SIP", NULL, &options);
2286 if (!sip_threadpool) {
2287 ast_log(LOG_ERROR, "Failed to create SIP threadpool. Aborting load\n");
2288 pj_pool_release(memory_pool);
2290 pjsip_endpt_destroy(ast_pjsip_endpoint);
2291 ast_pjsip_endpoint = NULL;
2292 pj_caching_pool_destroy(&caching_pool);
2293 return AST_MODULE_LOAD_DECLINE;
2296 pjsip_tsx_layer_init_module(ast_pjsip_endpoint);
2297 pjsip_ua_init_module(ast_pjsip_endpoint, NULL);
2299 monitor_continue = 1;
2300 status = pj_thread_create(memory_pool, "SIP", (pj_thread_proc *) &monitor_thread_exec,
2301 NULL, PJ_THREAD_DEFAULT_STACK_SIZE * 2, 0, &monitor_thread);
2302 if (status != PJ_SUCCESS) {
2303 ast_log(LOG_ERROR, "Failed to start SIP monitor thread. Aborting load\n");
2304 pj_pool_release(memory_pool);
2306 pjsip_endpt_destroy(ast_pjsip_endpoint);
2307 ast_pjsip_endpoint = NULL;
2308 pj_caching_pool_destroy(&caching_pool);
2309 return AST_MODULE_LOAD_DECLINE;
2312 ast_sip_initialize_global_headers();
2314 if (ast_res_pjsip_initialize_configuration(ast_module_info)) {
2315 ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
2316 ast_sip_destroy_global_headers();
2317 stop_monitor_thread();
2318 pj_pool_release(memory_pool);
2320 pjsip_endpt_destroy(ast_pjsip_endpoint);
2321 ast_pjsip_endpoint = NULL;
2322 pj_caching_pool_destroy(&caching_pool);
2323 return AST_MODULE_LOAD_DECLINE;
2326 if (ast_sip_initialize_distributor()) {
2327 ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n");
2328 ast_res_pjsip_destroy_configuration();
2329 ast_sip_destroy_global_headers();
2330 stop_monitor_thread();
2331 pj_pool_release(memory_pool);
2333 pjsip_endpt_destroy(ast_pjsip_endpoint);
2334 ast_pjsip_endpoint = NULL;
2335 pj_caching_pool_destroy(&caching_pool);
2336 return AST_MODULE_LOAD_DECLINE;
2339 if (ast_sip_register_service(&supplement_module)) {
2340 ast_log(LOG_ERROR, "Failed to initialize supplement hooks. Aborting load\n");
2341 ast_sip_destroy_distributor();
2342 ast_res_pjsip_destroy_configuration();
2343 ast_sip_destroy_global_headers();
2344 stop_monitor_thread();
2345 pj_pool_release(memory_pool);
2347 pjsip_endpt_destroy(ast_pjsip_endpoint);
2348 ast_pjsip_endpoint = NULL;
2349 pj_caching_pool_destroy(&caching_pool);
2350 return AST_MODULE_LOAD_DECLINE;
2353 if (ast_sip_initialize_outbound_authentication()) {
2354 ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n");
2355 ast_sip_unregister_service(&supplement_module);
2356 ast_sip_destroy_distributor();
2357 ast_res_pjsip_destroy_configuration();
2358 ast_sip_destroy_global_headers();
2359 stop_monitor_thread();
2360 pj_pool_release(memory_pool);
2362 pjsip_endpt_destroy(ast_pjsip_endpoint);
2363 ast_pjsip_endpoint = NULL;
2364 pj_caching_pool_destroy(&caching_pool);
2365 return AST_MODULE_LOAD_DECLINE;
2368 ast_res_pjsip_init_options_handling(0);
2370 ast_module_ref(ast_module_info->self);
2372 return AST_MODULE_LOAD_SUCCESS;
2375 static int reload_module(void)
2377 if (ast_res_pjsip_reload_configuration()) {
2378 return AST_MODULE_LOAD_DECLINE;
2380 ast_res_pjsip_init_options_handling(1);
2384 static int unload_module(void)
2386 /* This will never get called as this module can't be unloaded */
2390 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Basic SIP resource",
2391 .load = load_module,
2392 .unload = unload_module,
2393 .reload = reload_module,
2394 .load_pri = AST_MODPRI_CHANNEL_DEPEND - 5,
projects / asterisk/asterisk.git / blob