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>
900 <configOption name="user_agent">
901 <synopsis>User-Agent header from registration.</synopsis>
903 The User-Agent is automatically stored based on data present in incoming SIP
904 REGISTER requests and is not intended to be configured manually.
905 </para></description>
908 <configObject name="aor">
909 <synopsis>The configuration for a location of an endpoint</synopsis>
911 An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
912 AoRs are specified, an endpoint will not be reachable by Asterisk.
913 Beyond that, an AoR has other uses within Asterisk, such as inbound
916 An <literal>AoR</literal> is a way to allow dialing a group
917 of <literal>Contacts</literal> that all use the same
918 <literal>endpoint</literal> for calls.
920 This can be used as another way of grouping a list of contacts to dial
921 rather than specifing them each directly when dialing via the dialplan.
922 This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
924 Registrations: For Asterisk to match an inbound registration to an endpoint,
925 the AoR object name must match the user portion of the SIP URI in the "To:"
926 header of the inbound SIP registration. That will usually be equivalent
927 to the "user name" set in your hard or soft phones configuration.
928 </para></description>
929 <configOption name="contact">
930 <synopsis>Permanent contacts assigned to AoR</synopsis>
932 Contacts specified will be called whenever referenced
933 by <literal>chan_pjsip</literal>.
935 Use a separate "contact=" entry for each contact required. Contacts
936 are specified using a SIP URI.
937 </para></description>
939 <configOption name="default_expiration" default="3600">
940 <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
942 <configOption name="mailboxes">
943 <synopsis>Allow subscriptions for the specified mailbox(es)</synopsis>
944 <description><para>This option applies when an external entity subscribes to an AoR
945 for Message Waiting Indications. The mailboxes specified will be subscribed to.
946 More than one mailbox can be specified with a comma-delimited string.
947 app_voicemail mailboxes must be specified as mailbox@context;
948 for example: mailboxes=6001@default. For mailboxes provided by external sources,
949 such as through the res_external_mwi module, you must specify strings supported by
952 For endpoints that cannot SUBSCRIBE for MWI, you can set the <literal>mailboxes</literal> option in your
953 endpoint configuration section to enable unsolicited MWI NOTIFYs to the endpoint.
954 </para></description>
956 <configOption name="maximum_expiration" default="7200">
957 <synopsis>Maximum time to keep an AoR</synopsis>
959 Maximium time to keep a peer with explicit expiration. Time in seconds.
960 </para></description>
962 <configOption name="max_contacts" default="0">
963 <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
965 Maximum number of contacts that can associate with this AoR. This value does
966 not affect the number of contacts that can be added with the "contact" option.
967 It only limits contacts added through external interaction, such as
970 <note><para>This should be set to <literal>1</literal> and
971 <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
972 wish to stick with the older <literal>chan_sip</literal> behaviour.
976 <configOption name="minimum_expiration" default="60">
977 <synopsis>Minimum keep alive time for an AoR</synopsis>
979 Minimum time to keep a peer with an explict expiration. Time in seconds.
980 </para></description>
982 <configOption name="remove_existing" default="no">
983 <synopsis>Determines whether new contacts replace existing ones.</synopsis>
985 On receiving a new registration to the AoR should it remove
986 the existing contact that was registered against it?
988 <note><para>This should be set to <literal>yes</literal> and
989 <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
990 wish to stick with the older <literal>chan_sip</literal> behaviour.
994 <configOption name="type">
995 <synopsis>Must be of type 'aor'.</synopsis>
997 <configOption name="qualify_frequency" default="0">
998 <synopsis>Interval at which to qualify an AoR</synopsis>
1000 Interval between attempts to qualify the AoR for reachability.
1001 If <literal>0</literal> never qualify. Time in seconds.
1002 </para></description>
1004 <configOption name="authenticate_qualify" default="no">
1005 <synopsis>Authenticates a qualify request if needed</synopsis>
1007 If true and a qualify request receives a challenge or authenticate response
1008 authentication is attempted before declaring the contact available.
1009 </para></description>
1011 <configOption name="outbound_proxy">
1012 <synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
1014 If set the provided URI will be used as the outbound proxy when an
1015 OPTIONS request is sent to a contact for qualify purposes.
1016 </para></description>
1018 <configOption name="support_path">
1019 <synopsis>Enables Path support for REGISTER requests and Route support for other requests.</synopsis>
1021 When this option is enabled, the Path headers in register requests will be saved
1022 and its contents will be used in Route headers for outbound out-of-dialog requests
1023 and in Path headers for outbound 200 responses. Path support will also be indicated
1024 in the Supported header.
1025 </para></description>
1028 <configObject name="system">
1029 <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
1031 The settings in this section are global. In addition to being global, the values will
1032 not be re-evaluated when a reload is performed. This is because the values must be set
1033 before the SIP stack is initialized. The only way to reset these values is to either
1034 restart Asterisk, or unload res_pjsip.so and then load it again.
1035 </para></description>
1036 <configOption name="timer_t1" default="500">
1037 <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
1039 Timer T1 is the base for determining how long to wait before retransmitting
1040 requests that receive no response when using an unreliable transport (e.g. UDP).
1041 For more information on this timer, see RFC 3261, Section 17.1.1.1.
1042 </para></description>
1044 <configOption name="timer_b" default="32000">
1045 <synopsis>Set transaction timer B value (milliseconds).</synopsis>
1047 Timer B determines the maximum amount of time to wait after sending an INVITE
1048 request before terminating the transaction. It is recommended that this be set
1049 to 64 * Timer T1, but it may be set higher if desired. For more information on
1050 this timer, see RFC 3261, Section 17.1.1.1.
1051 </para></description>
1053 <configOption name="compact_headers" default="no">
1054 <synopsis>Use the short forms of common SIP header names.</synopsis>
1056 <configOption name="threadpool_initial_size" default="0">
1057 <synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
1059 <configOption name="threadpool_auto_increment" default="5">
1060 <synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
1062 <configOption name="threadpool_idle_timeout" default="60">
1063 <synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
1065 <configOption name="threadpool_max_size" default="0">
1066 <synopsis>Maximum number of threads in the res_pjsip threadpool.
1067 A value of 0 indicates no maximum.</synopsis>
1069 <configOption name="type">
1070 <synopsis>Must be of type 'system'.</synopsis>
1073 <configObject name="global">
1074 <synopsis>Options that apply globally to all SIP communications</synopsis>
1076 The settings in this section are global. Unlike options in the <literal>system</literal>
1077 section, these options can be refreshed by performing a reload.
1078 </para></description>
1079 <configOption name="max_forwards" default="70">
1080 <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
1082 <configOption name="type">
1083 <synopsis>Must be of type 'global'.</synopsis>
1085 <configOption name="user_agent" default="Asterisk <Asterisk Version>">
1086 <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
1088 <configOption name="default_outbound_endpoint" default="default_outbound_endpoint">
1089 <synopsis>Endpoint to use when sending an outbound request to a URI without a specified endpoint.</synopsis>
1091 <configOption name="debug" default="no">
1092 <synopsis>Enable/Disable SIP debug logging. Valid options include yes|no or
1093 a host address</synopsis>
1098 <manager name="PJSIPQualify" language="en_US">
1100 Qualify a chan_pjsip endpoint.
1103 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1104 <parameter name="Endpoint" required="true">
1105 <para>The endpoint you want to qualify.</para>
1109 <para>Qualify a chan_pjsip endpoint.</para>
1112 <manager name="PJSIPShowEndpoints" language="en_US">
1114 Lists PJSIP endpoints.
1119 Provides a listing of all endpoints. For each endpoint an <literal>EndpointList</literal> event
1120 is raised that contains relevant attributes and status information. Once all
1121 endpoints have been listed an <literal>EndpointListComplete</literal> event is issued.
1125 <manager name="PJSIPShowEndpoint" language="en_US">
1127 Detail listing of an endpoint and its objects.
1130 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1131 <parameter name="Endpoint" required="true">
1132 <para>The endpoint to list.</para>
1137 Provides a detailed listing of options for a given endpoint. Events are issued
1138 showing the configuration and status of the endpoint and associated objects. These
1139 events include <literal>EndpointDetail</literal>, <literal>AorDetail</literal>,
1140 <literal>AuthDetail</literal>, <literal>TransportDetail</literal>, and
1141 <literal>IdentifyDetail</literal>. Some events may be listed multiple times if multiple objects are
1142 associated (for instance AoRs). Once all detail events have been raised a final
1143 <literal>EndpointDetailComplete</literal> event is issued.
1149 #define MOD_DATA_CONTACT "contact"
1151 static pjsip_endpoint *ast_pjsip_endpoint;
1153 static struct ast_threadpool *sip_threadpool;
1155 static int register_service(void *data)
1157 pjsip_module **module = data;
1158 if (!ast_pjsip_endpoint) {
1159 ast_log(LOG_ERROR, "There is no PJSIP endpoint. Unable to register services\n");
1162 if (pjsip_endpt_register_module(ast_pjsip_endpoint, *module) != PJ_SUCCESS) {
1163 ast_log(LOG_ERROR, "Unable to register module %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1166 ast_debug(1, "Registered SIP service %.*s (%p)\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name), *module);
1167 ast_module_ref(ast_module_info->self);
1171 int ast_sip_register_service(pjsip_module *module)
1173 return ast_sip_push_task_synchronous(NULL, register_service, &module);
1176 static int unregister_service(void *data)
1178 pjsip_module **module = data;
1179 ast_module_unref(ast_module_info->self);
1180 if (!ast_pjsip_endpoint) {
1183 pjsip_endpt_unregister_module(ast_pjsip_endpoint, *module);
1184 ast_debug(1, "Unregistered SIP service %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1188 void ast_sip_unregister_service(pjsip_module *module)
1190 ast_sip_push_task_synchronous(NULL, unregister_service, &module);
1193 static struct ast_sip_authenticator *registered_authenticator;
1195 int ast_sip_register_authenticator(struct ast_sip_authenticator *auth)
1197 if (registered_authenticator) {
1198 ast_log(LOG_WARNING, "Authenticator %p is already registered. Cannot register a new one\n", registered_authenticator);
1201 registered_authenticator = auth;
1202 ast_debug(1, "Registered SIP authenticator module %p\n", auth);
1203 ast_module_ref(ast_module_info->self);
1207 void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth)
1209 if (registered_authenticator != auth) {
1210 ast_log(LOG_WARNING, "Trying to unregister authenticator %p but authenticator %p registered\n",
1211 auth, registered_authenticator);
1214 registered_authenticator = NULL;
1215 ast_debug(1, "Unregistered SIP authenticator %p\n", auth);
1216 ast_module_unref(ast_module_info->self);
1219 int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1221 if (!registered_authenticator) {
1222 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is not required\n");
1226 return registered_authenticator->requires_authentication(endpoint, rdata);
1229 enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
1230 pjsip_rx_data *rdata, pjsip_tx_data *tdata)
1232 if (!registered_authenticator) {
1233 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is successful\n");
1236 return registered_authenticator->check_authentication(endpoint, rdata, tdata);
1239 static struct ast_sip_outbound_authenticator *registered_outbound_authenticator;
1241 int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1243 if (registered_outbound_authenticator) {
1244 ast_log(LOG_WARNING, "Outbound authenticator %p is already registered. Cannot register a new one\n", registered_outbound_authenticator);
1247 registered_outbound_authenticator = auth;
1248 ast_debug(1, "Registered SIP outbound authenticator module %p\n", auth);
1249 ast_module_ref(ast_module_info->self);
1253 void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1255 if (registered_outbound_authenticator != auth) {
1256 ast_log(LOG_WARNING, "Trying to unregister outbound authenticator %p but outbound authenticator %p registered\n",
1257 auth, registered_outbound_authenticator);
1260 registered_outbound_authenticator = NULL;
1261 ast_debug(1, "Unregistered SIP outbound authenticator %p\n", auth);
1262 ast_module_unref(ast_module_info->self);
1265 int ast_sip_create_request_with_auth(const struct ast_sip_auth_vector *auths, pjsip_rx_data *challenge,
1266 pjsip_transaction *tsx, pjsip_tx_data **new_request)
1268 if (!registered_outbound_authenticator) {
1269 ast_log(LOG_WARNING, "No SIP outbound authenticator registered. Cannot respond to authentication challenge\n");
1272 return registered_outbound_authenticator->create_request_with_auth(auths, challenge, tsx, new_request);
1275 struct endpoint_identifier_list {
1276 struct ast_sip_endpoint_identifier *identifier;
1277 AST_RWLIST_ENTRY(endpoint_identifier_list) list;
1280 static AST_RWLIST_HEAD_STATIC(endpoint_identifiers, endpoint_identifier_list);
1282 int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1284 struct endpoint_identifier_list *id_list_item;
1285 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1287 id_list_item = ast_calloc(1, sizeof(*id_list_item));
1288 if (!id_list_item) {
1289 ast_log(LOG_ERROR, "Unabled to add endpoint identifier. Out of memory.\n");
1292 id_list_item->identifier = identifier;
1294 AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list);
1295 ast_debug(1, "Registered endpoint identifier %p\n", identifier);
1297 ast_module_ref(ast_module_info->self);
1301 void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1303 struct endpoint_identifier_list *iter;
1304 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1305 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) {
1306 if (iter->identifier == identifier) {
1307 AST_RWLIST_REMOVE_CURRENT(list);
1309 ast_debug(1, "Unregistered endpoint identifier %p\n", identifier);
1310 ast_module_unref(ast_module_info->self);
1314 AST_RWLIST_TRAVERSE_SAFE_END;
1317 struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata)
1319 struct endpoint_identifier_list *iter;
1320 struct ast_sip_endpoint *endpoint = NULL;
1321 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1322 AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) {
1323 ast_assert(iter->identifier->identify_endpoint != NULL);
1324 endpoint = iter->identifier->identify_endpoint(rdata);
1332 AST_RWLIST_HEAD_STATIC(endpoint_formatters, ast_sip_endpoint_formatter);
1334 int ast_sip_register_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
1336 SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1337 AST_RWLIST_INSERT_TAIL(&endpoint_formatters, obj, next);
1338 ast_module_ref(ast_module_info->self);
1342 void ast_sip_unregister_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
1344 struct ast_sip_endpoint_formatter *i;
1345 SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1346 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_formatters, i, next) {
1348 AST_RWLIST_REMOVE_CURRENT(next);
1349 ast_module_unref(ast_module_info->self);
1353 AST_RWLIST_TRAVERSE_SAFE_END;
1356 int ast_sip_format_endpoint_ami(struct ast_sip_endpoint *endpoint,
1357 struct ast_sip_ami *ami, int *count)
1360 struct ast_sip_endpoint_formatter *i;
1361 SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1363 AST_RWLIST_TRAVERSE(&endpoint_formatters, i, next) {
1364 if (i->format_ami && ((res = i->format_ami(endpoint, ami)) < 0)) {
1375 pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
1377 return ast_pjsip_endpoint;
1380 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)
1382 pj_str_t tmp, local_addr;
1384 pjsip_sip_uri *sip_uri;
1385 pjsip_transport_type_e type = PJSIP_TRANSPORT_UNSPECIFIED;
1387 char uuid_str[AST_UUID_STR_LEN];
1389 if (ast_strlen_zero(user)) {
1390 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1394 user = ast_uuid_to_str(uuid, uuid_str, sizeof(uuid_str));
1397 /* Parse the provided target URI so we can determine what transport it will end up using */
1398 pj_strdup_with_null(pool, &tmp, target);
1400 if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) ||
1401 (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
1405 sip_uri = pjsip_uri_get_uri(uri);
1407 /* Determine the transport type to use */
1408 if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
1409 type = PJSIP_TRANSPORT_TLS;
1410 } else if (!sip_uri->transport_param.slen) {
1411 type = PJSIP_TRANSPORT_UDP;
1413 type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
1416 if (type == PJSIP_TRANSPORT_UNSPECIFIED) {
1420 /* If the host is IPv6 turn the transport into an IPv6 version */
1421 if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1422 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1425 if (!ast_strlen_zero(domain)) {
1426 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1427 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1431 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1432 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1436 /* Get the local bound address for the transport that will be used when communicating with the provided URI */
1437 if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), pool, type, selector,
1438 &local_addr, &local_port) != PJ_SUCCESS) {
1440 /* If no local address can be retrieved using the transport manager use the host one */
1441 pj_strdup(pool, &local_addr, pj_gethostname());
1442 local_port = pjsip_transport_get_default_port_for_type(PJSIP_TRANSPORT_UDP);
1445 /* If IPv6 was specified in the transport, set the proper type */
1446 if (pj_strchr(&local_addr, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1447 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1450 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1451 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1452 "<sip:%s@%s%.*s%s:%d%s%s>",
1454 (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1455 (int)local_addr.slen,
1457 (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1459 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1460 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1465 static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
1467 RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
1468 const char *transport_name = endpoint->transport;
1470 if (ast_strlen_zero(transport_name)) {
1474 transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name);
1476 if (!transport || !transport->state) {
1477 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport '%s' for endpoint '%s'\n",
1478 transport_name, ast_sorcery_object_get_id(endpoint));
1482 if (transport->state->transport) {
1483 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1484 selector->u.transport = transport->state->transport;
1485 } else if (transport->state->factory) {
1486 selector->type = PJSIP_TPSELECTOR_LISTENER;
1487 selector->u.listener = transport->state->factory;
1495 pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
1497 char enclosed_uri[PJSIP_MAX_URL_SIZE];
1498 pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri, target_uri;
1499 pjsip_dialog *dlg = NULL;
1500 const char *outbound_proxy = endpoint->outbound_proxy;
1501 pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1502 static const pj_str_t HCONTACT = { "Contact", 7 };
1504 snprintf(enclosed_uri, sizeof(enclosed_uri), "<%s>", uri);
1505 pj_cstr(&remote_uri, enclosed_uri);
1507 pj_cstr(&target_uri, uri);
1509 if (pjsip_dlg_create_uac(pjsip_ua_instance(), &local_uri, NULL, &remote_uri, &target_uri, &dlg) != PJ_SUCCESS) {
1513 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1514 pjsip_dlg_terminate(dlg);
1518 if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
1519 pjsip_dlg_terminate(dlg);
1523 /* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
1524 pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
1525 dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
1526 dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
1528 /* If a request user has been specified and we are permitted to change it, do so */
1529 if (!ast_strlen_zero(request_user)) {
1530 pjsip_sip_uri *sip_uri;
1532 if (PJSIP_URI_SCHEME_IS_SIP(dlg->target) || PJSIP_URI_SCHEME_IS_SIPS(dlg->target)) {
1533 sip_uri = pjsip_uri_get_uri(dlg->target);
1534 pj_strdup2(dlg->pool, &sip_uri->user, request_user);
1536 if (PJSIP_URI_SCHEME_IS_SIP(dlg->remote.info->uri) || PJSIP_URI_SCHEME_IS_SIPS(dlg->remote.info->uri)) {
1537 sip_uri = pjsip_uri_get_uri(dlg->remote.info->uri);
1538 pj_strdup2(dlg->pool, &sip_uri->user, request_user);
1542 /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
1545 pjsip_dlg_set_transport(dlg, &selector);
1547 if (!ast_strlen_zero(outbound_proxy)) {
1548 pjsip_route_hdr route_set, *route;
1549 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1552 pj_list_init(&route_set);
1554 pj_strdup2_with_null(dlg->pool, &tmp, outbound_proxy);
1555 if (!(route = pjsip_parse_hdr(dlg->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1557 pjsip_dlg_terminate(dlg);
1560 pj_list_insert_nodes_before(&route_set, route);
1562 pjsip_dlg_set_route_set(dlg, &route_set);
1570 pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1574 pjsip_transport_type_e type = rdata->tp_info.transport->key.type;
1577 contact.ptr = pj_pool_alloc(rdata->tp_info.pool, PJSIP_MAX_URL_SIZE);
1578 contact.slen = pj_ansi_snprintf(contact.ptr, PJSIP_MAX_URL_SIZE,
1579 "<sip:%s%.*s%s:%d%s%s>",
1580 (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1581 (int)rdata->tp_info.transport->local_name.host.slen,
1582 rdata->tp_info.transport->local_name.host.ptr,
1583 (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1584 rdata->tp_info.transport->local_name.port,
1585 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1586 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1588 status = pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, &contact, &dlg);
1589 if (status != PJ_SUCCESS) {
1590 char err[PJ_ERR_MSG_SIZE];
1592 pj_strerror(status, err, sizeof(err));
1593 ast_log(LOG_ERROR, "Could not create dialog with endpoint %s. %s\n",
1594 ast_sorcery_object_get_id(endpoint), err);
1601 /* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
1602 static const pjsip_method info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
1603 static const pjsip_method message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
1607 const pjsip_method *pmethod;
1609 { "INVITE", &pjsip_invite_method },
1610 { "CANCEL", &pjsip_cancel_method },
1611 { "ACK", &pjsip_ack_method },
1612 { "BYE", &pjsip_bye_method },
1613 { "REGISTER", &pjsip_register_method },
1614 { "OPTIONS", &pjsip_options_method },
1615 { "SUBSCRIBE", &pjsip_subscribe_method },
1616 { "NOTIFY", &pjsip_notify_method },
1617 { "PUBLISH", &pjsip_publish_method },
1618 { "INFO", &info_method },
1619 { "MESSAGE", &message_method },
1622 static const pjsip_method *get_pjsip_method(const char *method)
1625 for (i = 0; i < ARRAY_LEN(methods); ++i) {
1626 if (!strcmp(method, methods[i].method)) {
1627 return methods[i].pmethod;
1633 static int create_in_dialog_request(const pjsip_method *method, struct pjsip_dialog *dlg, pjsip_tx_data **tdata)
1635 if (pjsip_dlg_create_request(dlg, method, -1, tdata) != PJ_SUCCESS) {
1636 ast_log(LOG_WARNING, "Unable to create in-dialog request.\n");
1643 static pj_bool_t supplement_on_rx_request(pjsip_rx_data *rdata);
1644 static pjsip_module supplement_module = {
1645 .name = { "Out of dialog supplement hook", 29 },
1647 .priority = PJSIP_MOD_PRIORITY_APPLICATION - 1,
1648 .on_rx_request = supplement_on_rx_request,
1651 static int create_out_of_dialog_request(const pjsip_method *method, struct ast_sip_endpoint *endpoint,
1652 const char *uri, struct ast_sip_contact *provided_contact, pjsip_tx_data **tdata)
1654 RAII_VAR(struct ast_sip_contact *, contact, ao2_bump(provided_contact), ao2_cleanup);
1655 pj_str_t remote_uri;
1658 pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1660 if (ast_strlen_zero(uri)) {
1661 if (!endpoint && !contact) {
1662 ast_log(LOG_ERROR, "An endpoint and/or uri must be specified\n");
1667 contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
1669 if (!contact || ast_strlen_zero(contact->uri)) {
1670 ast_log(LOG_ERROR, "Unable to retrieve contact for endpoint %s\n",
1671 ast_sorcery_object_get_id(endpoint));
1675 pj_cstr(&remote_uri, contact->uri);
1677 pj_cstr(&remote_uri, uri);
1681 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1682 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
1683 ast_sorcery_object_get_id(endpoint));
1688 pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
1691 ast_log(LOG_ERROR, "Unable to create PJLIB memory pool\n");
1695 if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL,
1696 endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
1697 ast_log(LOG_ERROR, "Unable to create From header for %.*s request to endpoint %s\n",
1698 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1699 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1703 if (pjsip_endpt_create_request(ast_sip_get_pjsip_endpoint(), method, &remote_uri,
1704 &from, &remote_uri, &from, NULL, -1, NULL, tdata) != PJ_SUCCESS) {
1705 ast_log(LOG_ERROR, "Unable to create outbound %.*s request to endpoint %s\n",
1706 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1707 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1711 /* If an outbound proxy is specified on the endpoint apply it to this request */
1712 if (endpoint && !ast_strlen_zero(endpoint->outbound_proxy) &&
1713 ast_sip_set_outbound_proxy((*tdata), endpoint->outbound_proxy)) {
1714 ast_log(LOG_ERROR, "Unable to apply outbound proxy on request %.*s to endpoint %s\n",
1715 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1716 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1720 ast_sip_mod_data_set((*tdata)->pool, (*tdata)->mod_data, supplement_module.id, MOD_DATA_CONTACT, ao2_bump(contact));
1722 /* We can release this pool since request creation copied all the necessary
1723 * data into the outbound request's pool
1725 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1729 int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
1730 struct ast_sip_endpoint *endpoint, const char *uri,
1731 struct ast_sip_contact *contact, pjsip_tx_data **tdata)
1733 const pjsip_method *pmethod = get_pjsip_method(method);
1736 ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
1741 return create_in_dialog_request(pmethod, dlg, tdata);
1743 return create_out_of_dialog_request(pmethod, endpoint, uri, contact, tdata);
1747 AST_RWLIST_HEAD_STATIC(supplements, ast_sip_supplement);
1749 int ast_sip_register_supplement(struct ast_sip_supplement *supplement)
1751 struct ast_sip_supplement *iter;
1753 SCOPED_LOCK(lock, &supplements, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1755 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&supplements, iter, next) {
1756 if (iter->priority > supplement->priority) {
1757 AST_RWLIST_INSERT_BEFORE_CURRENT(supplement, next);
1762 AST_RWLIST_TRAVERSE_SAFE_END;
1765 AST_RWLIST_INSERT_TAIL(&supplements, supplement, next);
1767 ast_module_ref(ast_module_info->self);
1771 void ast_sip_unregister_supplement(struct ast_sip_supplement *supplement)
1773 struct ast_sip_supplement *iter;
1774 SCOPED_LOCK(lock, &supplements, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1775 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&supplements, iter, next) {
1776 if (supplement == iter) {
1777 AST_RWLIST_REMOVE_CURRENT(next);
1778 ast_module_unref(ast_module_info->self);
1782 AST_RWLIST_TRAVERSE_SAFE_END;
1785 static int send_in_dialog_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg)
1787 if (pjsip_dlg_send_request(dlg, tdata, -1, NULL) != PJ_SUCCESS) {
1788 ast_log(LOG_WARNING, "Unable to send in-dialog request.\n");
1794 static pj_bool_t does_method_match(const pj_str_t *message_method, const char *supplement_method)
1798 if (ast_strlen_zero(supplement_method)) {
1802 pj_cstr(&method, supplement_method);
1804 return pj_stristr(&method, message_method) ? PJ_TRUE : PJ_FALSE;
1807 /*! \brief Structure to hold information about an outbound request */
1808 struct send_request_data {
1809 struct ast_sip_endpoint *endpoint; /*! The endpoint associated with this request */
1810 void *token; /*! Information to be provided to the callback upon receipt of a response */
1811 void (*callback)(void *token, pjsip_event *e); /*! The callback to be called upon receipt of a response */
1814 static void send_request_data_destroy(void *obj)
1816 struct send_request_data *req_data = obj;
1817 ao2_cleanup(req_data->endpoint);
1820 static struct send_request_data *send_request_data_alloc(struct ast_sip_endpoint *endpoint,
1821 void *token, void (*callback)(void *token, pjsip_event *e))
1823 struct send_request_data *req_data = ao2_alloc(sizeof(*req_data), send_request_data_destroy);
1829 req_data->endpoint = ao2_bump(endpoint);
1830 req_data->token = token;
1831 req_data->callback = callback;
1836 static void send_request_cb(void *token, pjsip_event *e)
1838 RAII_VAR(struct send_request_data *, req_data, token, ao2_cleanup);
1839 pjsip_transaction *tsx = e->body.tsx_state.tsx;
1840 pjsip_rx_data *challenge = e->body.tsx_state.src.rdata;
1841 pjsip_tx_data *tdata;
1842 struct ast_sip_supplement *supplement;
1844 AST_RWLIST_RDLOCK(&supplements);
1845 AST_LIST_TRAVERSE(&supplements, supplement, next) {
1846 if (supplement->incoming_response && does_method_match(&challenge->msg_info.cseq->method.name, supplement->method)) {
1847 supplement->incoming_response(req_data->endpoint, challenge);
1850 AST_RWLIST_UNLOCK(&supplements);
1852 if (tsx->status_code == 401 || tsx->status_code == 407) {
1853 if (!ast_sip_create_request_with_auth(&req_data->endpoint->outbound_auths, challenge, tsx, &tdata)) {
1854 pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, req_data->token, req_data->callback);
1859 if (req_data->callback) {
1860 req_data->callback(req_data->token, e);
1864 static int send_out_of_dialog_request(pjsip_tx_data *tdata, struct ast_sip_endpoint *endpoint,
1865 void *token, void (*callback)(void *token, pjsip_event *e))
1867 struct ast_sip_supplement *supplement;
1868 struct send_request_data *req_data = send_request_data_alloc(endpoint, token, callback);
1869 struct ast_sip_contact *contact = ast_sip_mod_data_get(tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT);
1875 AST_RWLIST_RDLOCK(&supplements);
1876 AST_LIST_TRAVERSE(&supplements, supplement, next) {
1877 if (supplement->outgoing_request && does_method_match(&tdata->msg->line.req.method.name, supplement->method)) {
1878 supplement->outgoing_request(endpoint, contact, tdata);
1881 AST_RWLIST_UNLOCK(&supplements);
1883 ast_sip_mod_data_set(tdata->pool, tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT, NULL);
1884 ao2_cleanup(contact);
1886 if (pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, req_data, send_request_cb) != PJ_SUCCESS) {
1887 ast_log(LOG_ERROR, "Error attempting to send outbound %.*s request to endpoint %s\n",
1888 (int) pj_strlen(&tdata->msg->line.req.method.name),
1889 pj_strbuf(&tdata->msg->line.req.method.name),
1890 ast_sorcery_object_get_id(endpoint));
1891 ao2_cleanup(req_data);
1898 int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg,
1899 struct ast_sip_endpoint *endpoint, void *token,
1900 void (*callback)(void *token, pjsip_event *e))
1902 ast_assert(tdata->msg->type == PJSIP_REQUEST_MSG);
1905 return send_in_dialog_request(tdata, dlg);
1907 return send_out_of_dialog_request(tdata, endpoint, token, callback);
1911 int ast_sip_set_outbound_proxy(pjsip_tx_data *tdata, const char *proxy)
1913 pjsip_route_hdr *route;
1914 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1917 pj_strdup2_with_null(tdata->pool, &tmp, proxy);
1918 if (!(route = pjsip_parse_hdr(tdata->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1922 pj_list_insert_nodes_before(&tdata->msg->hdr, (pjsip_hdr*)route);
1927 int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value)
1931 pjsip_generic_string_hdr *hdr;
1933 pj_cstr(&hdr_name, name);
1934 pj_cstr(&hdr_value, value);
1936 hdr = pjsip_generic_string_hdr_create(tdata->pool, &hdr_name, &hdr_value);
1938 pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) hdr);
1942 static pjsip_msg_body *ast_body_to_pjsip_body(pj_pool_t *pool, const struct ast_sip_body *body)
1948 pj_cstr(&type, body->type);
1949 pj_cstr(&subtype, body->subtype);
1950 pj_cstr(&body_text, body->body_text);
1952 return pjsip_msg_body_create(pool, &type, &subtype, &body_text);
1955 int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body)
1957 pjsip_msg_body *pjsip_body = ast_body_to_pjsip_body(tdata->pool, body);
1958 tdata->msg->body = pjsip_body;
1962 int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies)
1965 /* NULL for type and subtype automatically creates "multipart/mixed" */
1966 pjsip_msg_body *body = pjsip_multipart_create(tdata->pool, NULL, NULL);
1968 for (i = 0; i < num_bodies; ++i) {
1969 pjsip_multipart_part *part = pjsip_multipart_create_part(tdata->pool);
1970 part->body = ast_body_to_pjsip_body(tdata->pool, bodies[i]);
1971 pjsip_multipart_add_part(tdata->pool, body, part);
1974 tdata->msg->body = body;
1978 int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text)
1980 size_t combined_size = strlen(body_text) + tdata->msg->body->len;
1981 struct ast_str *body_buffer = ast_str_alloca(combined_size);
1983 ast_str_set(&body_buffer, 0, "%.*s%s", (int) tdata->msg->body->len, (char *) tdata->msg->body->data, body_text);
1985 tdata->msg->body->data = pj_pool_alloc(tdata->pool, combined_size);
1986 pj_memcpy(tdata->msg->body->data, ast_str_buffer(body_buffer), combined_size);
1987 tdata->msg->body->len = combined_size;
1992 struct ast_taskprocessor *ast_sip_create_serializer(void)
1994 struct ast_taskprocessor *serializer;
1995 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1996 char name[AST_UUID_STR_LEN];
2002 ast_uuid_to_str(uuid, name, sizeof(name));
2004 serializer = ast_threadpool_serializer(name, sip_threadpool);
2011 int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
2014 return ast_taskprocessor_push(serializer, sip_task, task_data);
2016 return ast_threadpool_push(sip_threadpool, sip_task, task_data);
2020 struct sync_task_data {
2025 int (*task)(void *);
2029 static int sync_task(void *data)
2031 struct sync_task_data *std = data;
2032 std->fail = std->task(std->task_data);
2034 ast_mutex_lock(&std->lock);
2036 ast_cond_signal(&std->cond);
2037 ast_mutex_unlock(&std->lock);
2041 int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
2043 /* This method is an onion */
2044 struct sync_task_data std;
2046 if (ast_sip_thread_is_servant()) {
2047 return sip_task(task_data);
2050 ast_mutex_init(&std.lock);
2051 ast_cond_init(&std.cond, NULL);
2052 std.fail = std.complete = 0;
2053 std.task = sip_task;
2054 std.task_data = task_data;
2057 if (ast_taskprocessor_push(serializer, sync_task, &std)) {
2061 if (ast_threadpool_push(sip_threadpool, sync_task, &std)) {
2066 ast_mutex_lock(&std.lock);
2067 while (!std.complete) {
2068 ast_cond_wait(&std.cond, &std.lock);
2070 ast_mutex_unlock(&std.lock);
2072 ast_mutex_destroy(&std.lock);
2073 ast_cond_destroy(&std.cond);
2077 void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
2079 size_t chars_to_copy = MIN(size - 1, pj_strlen(src));
2080 memcpy(dest, pj_strbuf(src), chars_to_copy);
2081 dest[chars_to_copy] = '\0';
2084 int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype)
2086 pjsip_media_type compare;
2088 if (!content_type) {
2092 pjsip_media_type_init2(&compare, type, subtype);
2094 return pjsip_media_type_cmp(content_type, &compare, 0) ? 0 : -1;
2097 pj_caching_pool caching_pool;
2098 pj_pool_t *memory_pool;
2099 pj_thread_t *monitor_thread;
2100 static int monitor_continue;
2102 static void *monitor_thread_exec(void *endpt)
2104 while (monitor_continue) {
2105 const pj_time_val delay = {0, 10};
2106 pjsip_endpt_handle_events(ast_pjsip_endpoint, &delay);
2111 static void stop_monitor_thread(void)
2113 monitor_continue = 0;
2114 pj_thread_join(monitor_thread);
2117 AST_THREADSTORAGE(pj_thread_storage);
2118 AST_THREADSTORAGE(servant_id_storage);
2119 #define SIP_SERVANT_ID 0x5E2F1D
2121 static void sip_thread_start(void)
2123 pj_thread_desc *desc;
2124 pj_thread_t *thread;
2125 uint32_t *servant_id;
2127 servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
2129 ast_log(LOG_ERROR, "Could not set SIP servant ID in thread-local storage.\n");
2132 *servant_id = SIP_SERVANT_ID;
2134 desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc));
2136 ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n");
2139 pj_bzero(*desc, sizeof(*desc));
2141 if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) {
2142 ast_log(LOG_ERROR, "Couldn't register thread with PJLIB.\n");
2146 int ast_sip_thread_is_servant(void)
2148 uint32_t *servant_id;
2150 servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
2155 return *servant_id == SIP_SERVANT_ID;
2158 void *ast_sip_dict_get(void *ht, const char *key)
2160 unsigned int hval = 0;
2166 return pj_hash_get(ht, key, PJ_HASH_KEY_STRING, &hval);
2169 void *ast_sip_dict_set(pj_pool_t* pool, void *ht,
2170 const char *key, void *val)
2173 ht = pj_hash_create(pool, 11);
2176 pj_hash_set(pool, ht, key, PJ_HASH_KEY_STRING, 0, val);
2181 static pj_bool_t supplement_on_rx_request(pjsip_rx_data *rdata)
2183 struct ast_sip_supplement *supplement;
2185 if (pjsip_rdata_get_dlg(rdata)) {
2189 AST_RWLIST_RDLOCK(&supplements);
2190 AST_LIST_TRAVERSE(&supplements, supplement, next) {
2191 if (supplement->incoming_request && does_method_match(&rdata->msg_info.msg->line.req.method.name, supplement->method)) {
2192 supplement->incoming_request(ast_pjsip_rdata_get_endpoint(rdata), rdata);
2195 AST_RWLIST_UNLOCK(&supplements);
2200 int ast_sip_send_response(pjsip_response_addr *res_addr, pjsip_tx_data *tdata, struct ast_sip_endpoint *sip_endpoint)
2202 struct ast_sip_supplement *supplement;
2203 pjsip_cseq_hdr *cseq = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_CSEQ, NULL);
2204 struct ast_sip_contact *contact = ast_sip_mod_data_get(tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT);
2206 AST_RWLIST_RDLOCK(&supplements);
2207 AST_LIST_TRAVERSE(&supplements, supplement, next) {
2208 if (supplement->outgoing_response && does_method_match(&cseq->method.name, supplement->method)) {
2209 supplement->outgoing_response(sip_endpoint, contact, tdata);
2212 AST_RWLIST_UNLOCK(&supplements);
2214 ast_sip_mod_data_set(tdata->pool, tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT, NULL);
2215 ao2_cleanup(contact);
2217 return pjsip_endpt_send_response(ast_sip_get_pjsip_endpoint(), res_addr, tdata, NULL, NULL);
2220 int ast_sip_create_response(const pjsip_rx_data *rdata, int st_code,
2221 struct ast_sip_contact *contact, pjsip_tx_data **tdata)
2223 int res = pjsip_endpt_create_response(ast_sip_get_pjsip_endpoint(), rdata, st_code, NULL, tdata);
2226 ast_sip_mod_data_set((*tdata)->pool, (*tdata)->mod_data, supplement_module.id, MOD_DATA_CONTACT, ao2_bump(contact));
2232 static void remove_request_headers(pjsip_endpoint *endpt)
2234 const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt);
2235 pjsip_hdr *iter = request_headers->next;
2237 while (iter != request_headers) {
2238 pjsip_hdr *to_erase = iter;
2240 pj_list_erase(to_erase);
2244 static int load_module(void)
2246 /* The third parameter is just copied from
2247 * example code from PJLIB. This can be adjusted
2251 struct ast_threadpool_options options;
2253 if (pj_init() != PJ_SUCCESS) {
2254 return AST_MODULE_LOAD_DECLINE;
2257 if (pjlib_util_init() != PJ_SUCCESS) {
2259 return AST_MODULE_LOAD_DECLINE;
2262 pj_caching_pool_init(&caching_pool, NULL, 1024 * 1024);
2263 if (pjsip_endpt_create(&caching_pool.factory, "SIP", &ast_pjsip_endpoint) != PJ_SUCCESS) {
2264 ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n");
2265 pj_caching_pool_destroy(&caching_pool);
2266 return AST_MODULE_LOAD_DECLINE;
2269 /* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that,
2270 * we need to stop PJSIP from doing it automatically
2272 remove_request_headers(ast_pjsip_endpoint);
2274 memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL);
2276 ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n");
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 if (ast_sip_initialize_system()) {
2284 ast_log(LOG_ERROR, "Failed to initialize SIP 'system' configuration section. Aborting load\n");
2285 pj_pool_release(memory_pool);
2287 pjsip_endpt_destroy(ast_pjsip_endpoint);
2288 ast_pjsip_endpoint = NULL;
2289 pj_caching_pool_destroy(&caching_pool);
2290 return AST_MODULE_LOAD_DECLINE;
2293 sip_get_threadpool_options(&options);
2294 options.thread_start = sip_thread_start;
2295 sip_threadpool = ast_threadpool_create("SIP", NULL, &options);
2296 if (!sip_threadpool) {
2297 ast_log(LOG_ERROR, "Failed to create SIP threadpool. Aborting load\n");
2298 pj_pool_release(memory_pool);
2300 pjsip_endpt_destroy(ast_pjsip_endpoint);
2301 ast_pjsip_endpoint = NULL;
2302 pj_caching_pool_destroy(&caching_pool);
2303 return AST_MODULE_LOAD_DECLINE;
2306 pjsip_tsx_layer_init_module(ast_pjsip_endpoint);
2307 pjsip_ua_init_module(ast_pjsip_endpoint, NULL);
2309 monitor_continue = 1;
2310 status = pj_thread_create(memory_pool, "SIP", (pj_thread_proc *) &monitor_thread_exec,
2311 NULL, PJ_THREAD_DEFAULT_STACK_SIZE * 2, 0, &monitor_thread);
2312 if (status != PJ_SUCCESS) {
2313 ast_log(LOG_ERROR, "Failed to start SIP monitor thread. Aborting load\n");
2314 pj_pool_release(memory_pool);
2316 pjsip_endpt_destroy(ast_pjsip_endpoint);
2317 ast_pjsip_endpoint = NULL;
2318 pj_caching_pool_destroy(&caching_pool);
2319 return AST_MODULE_LOAD_DECLINE;
2322 ast_sip_initialize_global_headers();
2324 if (ast_res_pjsip_initialize_configuration(ast_module_info)) {
2325 ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
2326 ast_sip_destroy_global_headers();
2327 stop_monitor_thread();
2328 pj_pool_release(memory_pool);
2330 pjsip_endpt_destroy(ast_pjsip_endpoint);
2331 ast_pjsip_endpoint = NULL;
2332 pj_caching_pool_destroy(&caching_pool);
2333 return AST_MODULE_LOAD_DECLINE;
2336 if (ast_sip_initialize_distributor()) {
2337 ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n");
2338 ast_res_pjsip_destroy_configuration();
2339 ast_sip_destroy_global_headers();
2340 stop_monitor_thread();
2341 pj_pool_release(memory_pool);
2343 pjsip_endpt_destroy(ast_pjsip_endpoint);
2344 ast_pjsip_endpoint = NULL;
2345 pj_caching_pool_destroy(&caching_pool);
2346 return AST_MODULE_LOAD_DECLINE;
2349 if (ast_sip_register_service(&supplement_module)) {
2350 ast_log(LOG_ERROR, "Failed to initialize supplement hooks. Aborting load\n");
2351 ast_sip_destroy_distributor();
2352 ast_res_pjsip_destroy_configuration();
2353 ast_sip_destroy_global_headers();
2354 stop_monitor_thread();
2355 pj_pool_release(memory_pool);
2357 pjsip_endpt_destroy(ast_pjsip_endpoint);
2358 ast_pjsip_endpoint = NULL;
2359 pj_caching_pool_destroy(&caching_pool);
2360 return AST_MODULE_LOAD_DECLINE;
2363 if (ast_sip_initialize_outbound_authentication()) {
2364 ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n");
2365 ast_sip_unregister_service(&supplement_module);
2366 ast_sip_destroy_distributor();
2367 ast_res_pjsip_destroy_configuration();
2368 ast_sip_destroy_global_headers();
2369 stop_monitor_thread();
2370 pj_pool_release(memory_pool);
2372 pjsip_endpt_destroy(ast_pjsip_endpoint);
2373 ast_pjsip_endpoint = NULL;
2374 pj_caching_pool_destroy(&caching_pool);
2375 return AST_MODULE_LOAD_DECLINE;
2378 ast_res_pjsip_init_options_handling(0);
2380 ast_module_ref(ast_module_info->self);
2382 return AST_MODULE_LOAD_SUCCESS;
2385 static int reload_module(void)
2387 if (ast_res_pjsip_reload_configuration()) {
2388 return AST_MODULE_LOAD_DECLINE;
2390 ast_res_pjsip_init_options_handling(1);
2394 static int unload_module(void)
2396 /* This will never get called as this module can't be unloaded */
2400 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Basic SIP resource",
2401 .load = load_module,
2402 .unload = unload_module,
2403 .reload = reload_module,
2404 .load_pri = AST_MODPRI_CHANNEL_DEPEND - 5,
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