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>Mailbox(es) to be associated with</synopsis>
271 <configOption name="moh_suggest" default="default">
272 <synopsis>Default Music On Hold class</synopsis>
274 <configOption name="outbound_auth">
275 <synopsis>Authentication object used for outbound requests</synopsis>
277 <configOption name="outbound_proxy">
278 <synopsis>Proxy through which to send requests, a full SIP URI must be provided</synopsis>
280 <configOption name="rewrite_contact">
281 <synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
283 On inbound SIP messages from this endpoint, the Contact header will be changed to have the
284 source IP address and port. This option does not affect outbound messages send to this
286 </para></description>
288 <configOption name="rtp_ipv6" default="no">
289 <synopsis>Allow use of IPv6 for RTP traffic</synopsis>
291 <configOption name="rtp_symmetric" default="no">
292 <synopsis>Enforce that RTP must be symmetric</synopsis>
294 <configOption name="send_diversion" default="yes">
295 <synopsis>Send the Diversion header, conveying the diversion
296 information to the called user agent</synopsis>
298 <configOption name="send_pai" default="no">
299 <synopsis>Send the P-Asserted-Identity header</synopsis>
301 <configOption name="send_rpid" default="no">
302 <synopsis>Send the Remote-Party-ID header</synopsis>
304 <configOption name="timers_min_se" default="90">
305 <synopsis>Minimum session timers expiration period</synopsis>
307 Minimium session timer expiration period. Time in seconds.
308 </para></description>
310 <configOption name="timers" default="yes">
311 <synopsis>Session timers for SIP packets</synopsis>
314 <enum name="forced" />
316 <enum name="required" />
321 <configOption name="timers_sess_expires" default="1800">
322 <synopsis>Maximum session timer expiration period</synopsis>
324 Maximium session timer expiration period. Time in seconds.
325 </para></description>
327 <configOption name="transport">
328 <synopsis>Desired transport configuration</synopsis>
330 This will set the desired transport configuration to send SIP data through.
332 <warning><para>Not specifying a transport will <emphasis>DEFAULT</emphasis>
333 to the first configured transport in <filename>pjsip.conf</filename> which is
334 valid for the URI we are trying to contact.
336 <warning><para>Transport configuration is not affected by reloads. In order to
337 change transports, a full Asterisk restart is required</para></warning>
340 <configOption name="trust_id_inbound" default="no">
341 <synopsis>Accept identification information received from this endpoint</synopsis>
342 <description><para>This option determines whether Asterisk will accept
343 identification from the endpoint from headers such as P-Asserted-Identity
344 or Remote-Party-ID header. This option applies both to calls originating from the
345 endpoint and calls originating from Asterisk. If <literal>no</literal>, the
346 configured Caller-ID from pjsip.conf will always be used as the identity for
347 the endpoint.</para></description>
349 <configOption name="trust_id_outbound" default="no">
350 <synopsis>Send private identification details to the endpoint.</synopsis>
351 <description><para>This option determines whether res_pjsip will send private
352 identification information to the endpoint. If <literal>no</literal>,
353 private Caller-ID information will not be forwarded to the endpoint.
354 "Private" in this case refers to any method of restricting identification.
355 Example: setting <replaceable>callerid_privacy</replaceable> to any
356 <literal>prohib</literal> variation.
357 Example: If <replaceable>trust_id_inbound</replaceable> is set to
358 <literal>yes</literal>, the presence of a <literal>Privacy: id</literal>
359 header in a SIP request or response would indicate the identification
360 provided in the request is private.</para></description>
362 <configOption name="type">
363 <synopsis>Must be of type 'endpoint'.</synopsis>
365 <configOption name="use_ptime" default="no">
366 <synopsis>Use Endpoint's requested packetisation interval</synopsis>
368 <configOption name="use_avpf" default="no">
369 <synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this
372 If set to <literal>yes</literal>, res_pjsip will use use the AVPF or SAVPF RTP
373 profile for all media offers on outbound calls and media updates and will
374 decline media offers not using the AVPF or SAVPF profile.
376 If set to <literal>no</literal>, res_pjsip will use use the AVP or SAVP RTP
377 profile for all media offers on outbound calls and media updates and will
378 decline media offers not using the AVP or SAVP profile.
379 </para></description>
381 <configOption name="media_encryption" default="no">
382 <synopsis>Determines whether res_pjsip will use and enforce usage of media encryption
383 for this endpoint.</synopsis>
386 <enum name="no"><para>
387 res_pjsip will offer no encryption and allow no encryption to be setup.
389 <enum name="sdes"><para>
390 res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP
391 transport should be used in conjunction with this option to prevent
392 exposure of media encryption keys.
394 <enum name="dtls"><para>
395 res_pjsip will offer DTLS-SRTP setup.
400 <configOption name="inband_progress" default="no">
401 <synopsis>Determines whether chan_pjsip will indicate ringing using inband
404 If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
405 when told to indicate ringing and will immediately start sending ringing
408 If set to <literal>no</literal>, chan_pjsip will send a 180 Ringing when told
409 to indicate ringing and will NOT send it as audio.
410 </para></description>
412 <configOption name="call_group">
413 <synopsis>The numeric pickup groups for a channel.</synopsis>
415 Can be set to a comma separated list of numbers or ranges between the values
416 of 0-63 (maximum of 64 groups).
417 </para></description>
419 <configOption name="pickup_group">
420 <synopsis>The numeric pickup groups that a channel can pickup.</synopsis>
422 Can be set to a comma separated list of numbers or ranges between the values
423 of 0-63 (maximum of 64 groups).
424 </para></description>
426 <configOption name="named_call_group">
427 <synopsis>The named pickup groups for a channel.</synopsis>
429 Can be set to a comma separated list of case sensitive strings limited by
430 supported line length.
431 </para></description>
433 <configOption name="named_pickup_group">
434 <synopsis>The named pickup groups that a channel can pickup.</synopsis>
436 Can be set to a comma separated list of case sensitive strings limited by
437 supported line length.
438 </para></description>
440 <configOption name="device_state_busy_at" default="0">
441 <synopsis>The number of in-use channels which will cause busy to be returned as device state</synopsis>
443 When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the
444 PJSIP channel driver will return busy as the device state instead of in use.
445 </para></description>
447 <configOption name="t38_udptl" default="no">
448 <synopsis>Whether T.38 UDPTL support is enabled or not</synopsis>
450 If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted
452 </para></description>
454 <configOption name="t38_udptl_ec" default="none">
455 <synopsis>T.38 UDPTL error correction method</synopsis>
458 <enum name="none"><para>
459 No error correction should be used.
461 <enum name="fec"><para>
462 Forward error correction should be used.
464 <enum name="redundancy"><para>
465 Redundacy error correction should be used.
470 <configOption name="t38_udptl_maxdatagram" default="0">
471 <synopsis>T.38 UDPTL maximum datagram size</synopsis>
473 This option can be set to override the maximum datagram of a remote endpoint for broken
475 </para></description>
477 <configOption name="fax_detect" default="no">
478 <synopsis>Whether CNG tone detection is enabled</synopsis>
480 This option can be set to send the session to the fax extension when a CNG tone is
482 </para></description>
484 <configOption name="t38_udptl_nat" default="no">
485 <synopsis>Whether NAT support is enabled on UDPTL sessions</synopsis>
487 When enabled the UDPTL stack will send UDPTL packets to the source address of
489 </para></description>
491 <configOption name="t38_udptl_ipv6" default="no">
492 <synopsis>Whether IPv6 is used for UDPTL Sessions</synopsis>
494 When enabled the UDPTL stack will use IPv6.
495 </para></description>
497 <configOption name="tone_zone">
498 <synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis>
500 <configOption name="language">
501 <synopsis>Set the default language to use for channels created for this endpoint.</synopsis>
503 <configOption name="one_touch_recording" default="no">
504 <synopsis>Determines whether one-touch recording is allowed for this endpoint.</synopsis>
506 <ref type="configOption">recordonfeature</ref>
507 <ref type="configOption">recordofffeature</ref>
510 <configOption name="record_on_feature" default="automixmon">
511 <synopsis>The feature to enact when one-touch recording is turned on.</synopsis>
513 <para>When an INFO request for one-touch recording arrives with a Record header set to "on", this
514 feature will be enabled for the channel. The feature designated here can be any built-in
515 or dynamic feature defined in features.conf.</para>
516 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
519 <ref type="configOption">one_touch_recording</ref>
520 <ref type="configOption">recordofffeature</ref>
523 <configOption name="record_off_feature" default="automixmon">
524 <synopsis>The feature to enact when one-touch recording is turned off.</synopsis>
526 <para>When an INFO request for one-touch recording arrives with a Record header set to "off", this
527 feature will be enabled for the channel. The feature designated here can be any built-in
528 or dynamic feature defined in features.conf.</para>
529 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
532 <ref type="configOption">one_touch_recording</ref>
533 <ref type="configOption">recordonfeature</ref>
536 <configOption name="rtp_engine" default="asterisk">
537 <synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis>
539 <configOption name="allow_transfer" default="yes">
540 <synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis>
542 <configOption name="sdp_owner" default="-">
543 <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
545 <configOption name="sdp_session" default="Asterisk">
546 <synopsis>String used for the SDP session (s=) line.</synopsis>
548 <configOption name="tos_audio">
549 <synopsis>DSCP TOS bits for audio streams</synopsis>
551 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
552 </para></description>
554 <configOption name="tos_video">
555 <synopsis>DSCP TOS bits for video streams</synopsis>
557 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
558 </para></description>
560 <configOption name="cos_audio">
561 <synopsis>Priority for audio streams</synopsis>
563 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
564 </para></description>
566 <configOption name="cos_video">
567 <synopsis>Priority for video streams</synopsis>
569 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
570 </para></description>
572 <configOption name="allow_subscribe" default="yes">
573 <synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis>
575 <configOption name="sub_min_expiry" default="60">
576 <synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis>
578 <configOption name="from_user">
579 <synopsis>Username to use in From header for requests to this endpoint.</synopsis>
581 <configOption name="mwi_from_user">
582 <synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis>
584 <configOption name="from_domain">
585 <synopsis>Domain to user in From header for requests to this endpoint.</synopsis>
587 <configOption name="dtls_verify">
588 <synopsis>Verify that the provided peer certificate is valid</synopsis>
590 This option only applies if <replaceable>media_encryption</replaceable> is
591 set to <literal>dtls</literal>.
592 </para></description>
594 <configOption name="dtls_rekey">
595 <synopsis>Interval at which to renegotiate the TLS session and rekey the SRTP session</synopsis>
597 This option only applies if <replaceable>media_encryption</replaceable> is
598 set to <literal>dtls</literal>.
600 If this is not set or the value provided is 0 rekeying will be disabled.
601 </para></description>
603 <configOption name="dtls_cert_file">
604 <synopsis>Path to certificate file to present to peer</synopsis>
606 This option only applies if <replaceable>media_encryption</replaceable> is
607 set to <literal>dtls</literal>.
608 </para></description>
610 <configOption name="dtls_private_key">
611 <synopsis>Path to private key for certificate file</synopsis>
613 This option only applies if <replaceable>media_encryption</replaceable> is
614 set to <literal>dtls</literal>.
615 </para></description>
617 <configOption name="dtls_cipher">
618 <synopsis>Cipher to use for DTLS negotiation</synopsis>
620 This option only applies if <replaceable>media_encryption</replaceable> is
621 set to <literal>dtls</literal>.
623 Many options for acceptable ciphers. See link for more:
624 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
625 </para></description>
627 <configOption name="dtls_ca_file">
628 <synopsis>Path to certificate authority certificate</synopsis>
630 This option only applies if <replaceable>media_encryption</replaceable> is
631 set to <literal>dtls</literal>.
632 </para></description>
634 <configOption name="dtls_ca_path">
635 <synopsis>Path to a directory containing certificate authority certificates</synopsis>
637 This option only applies if <replaceable>media_encryption</replaceable> is
638 set to <literal>dtls</literal>.
639 </para></description>
641 <configOption name="dtls_setup">
642 <synopsis>Whether we are willing to accept connections, connect to the other party, or both.</synopsis>
645 This option only applies if <replaceable>media_encryption</replaceable> is
646 set to <literal>dtls</literal>.
649 <enum name="active"><para>
650 res_pjsip will make a connection to the peer.
652 <enum name="passive"><para>
653 res_pjsip will accept connections from the peer.
655 <enum name="actpass"><para>
656 res_pjsip will offer and accept connections from the peer.
661 <configOption name="srtp_tag_32">
662 <synopsis>Determines whether 32 byte tags should be used instead of 80 byte tags.</synopsis>
664 This option only applies if <replaceable>media_encryption</replaceable> is
665 set to <literal>sdes</literal> or <literal>dtls</literal>.
666 </para></description>
668 <configOption name="set_var">
669 <synopsis>Variable set on a channel involving the endpoint.</synopsis>
671 When a new channel is created using the endpoint set the specified
672 variable(s) on that channel. For multiple channel variables specify
673 multiple 'set_var'(s).
674 </para></description>
677 <configObject name="auth">
678 <synopsis>Authentication type</synopsis>
680 Authentication objects hold the authentication information for use
681 by other objects such as <literal>endpoints</literal> or <literal>registrations</literal>.
682 This also allows for multiple objects to use a single auth object. See
683 the <literal>auth_type</literal> config option for password style choices.
684 </para></description>
685 <configOption name="auth_type" default="userpass">
686 <synopsis>Authentication type</synopsis>
688 This option specifies which of the password style config options should be read
689 when trying to authenticate an endpoint inbound request. If set to <literal>userpass</literal>
690 then we'll read from the 'password' option. For <literal>md5</literal> we'll read
695 <enum name="userpass"/>
699 <configOption name="nonce_lifetime" default="32">
700 <synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
702 <configOption name="md5_cred">
703 <synopsis>MD5 Hash used for authentication.</synopsis>
704 <description><para>Only used when auth_type is <literal>md5</literal>.</para></description>
706 <configOption name="password">
707 <synopsis>PlainText password used for authentication.</synopsis>
708 <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
710 <configOption name="realm" default="asterisk">
711 <synopsis>SIP realm for endpoint</synopsis>
713 <configOption name="type">
714 <synopsis>Must be 'auth'</synopsis>
716 <configOption name="username">
717 <synopsis>Username to use for account</synopsis>
720 <configObject name="domain_alias">
721 <synopsis>Domain Alias</synopsis>
723 Signifies that a domain is an alias. If the domain on a session is
724 not found to match an AoR then this object is used to see if we have
725 an alias for the AoR to which the endpoint is binding. This objects
726 name as defined in configuration should be the domain alias and a
727 config option is provided to specify the domain to be aliased.
728 </para></description>
729 <configOption name="type">
730 <synopsis>Must be of type 'domain_alias'.</synopsis>
732 <configOption name="domain">
733 <synopsis>Domain to be aliased</synopsis>
736 <configObject name="transport">
737 <synopsis>SIP Transport</synopsis>
739 <emphasis>Transports</emphasis>
741 <para>There are different transports and protocol derivatives
742 supported by <literal>res_pjsip</literal>. They are in order of
743 preference: UDP, TCP, and WebSocket (WS).</para>
744 <note><para>Changes to transport configuration in pjsip.conf will only be
745 effected on a complete restart of Asterisk. A module reload
746 will not suffice.</para></note>
748 <configOption name="async_operations" default="1">
749 <synopsis>Number of simultaneous Asynchronous Operations</synopsis>
751 <configOption name="bind">
752 <synopsis>IP Address and optional port to bind to for this transport</synopsis>
754 <configOption name="ca_list_file">
755 <synopsis>File containing a list of certificates to read (TLS ONLY)</synopsis>
757 <configOption name="cert_file">
758 <synopsis>Certificate file for endpoint (TLS ONLY)</synopsis>
760 <configOption name="cipher">
761 <synopsis>Preferred Cryptography Cipher (TLS ONLY)</synopsis>
763 Many options for acceptable ciphers see link for more:
764 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
765 </para></description>
767 <configOption name="domain">
768 <synopsis>Domain the transport comes from</synopsis>
770 <configOption name="external_media_address">
771 <synopsis>External IP address to use in RTP handling</synopsis>
773 When a request or response is sent out, if the destination of the
774 message is outside the IP network defined in the option <literal>localnet</literal>,
775 and the media address in the SDP is within the localnet network, then the
776 media address in the SDP will be rewritten to the value defined for
777 <literal>external_media_address</literal>.
778 </para></description>
780 <configOption name="external_signaling_address">
781 <synopsis>External address for SIP signalling</synopsis>
783 <configOption name="external_signaling_port" default="0">
784 <synopsis>External port for SIP signalling</synopsis>
786 <configOption name="method">
787 <synopsis>Method of SSL transport (TLS ONLY)</synopsis>
790 <enum name="default" />
791 <enum name="unspecified" />
792 <enum name="tlsv1" />
793 <enum name="sslv2" />
794 <enum name="sslv3" />
795 <enum name="sslv23" />
799 <configOption name="local_net">
800 <synopsis>Network to consider local (used for NAT purposes).</synopsis>
801 <description><para>This must be in CIDR or dotted decimal format with the IP
802 and mask separated with a slash ('/').</para></description>
804 <configOption name="password">
805 <synopsis>Password required for transport</synopsis>
807 <configOption name="priv_key_file">
808 <synopsis>Private key file (TLS ONLY)</synopsis>
810 <configOption name="protocol" default="udp">
811 <synopsis>Protocol to use for SIP traffic</synopsis>
822 <configOption name="require_client_cert" default="false">
823 <synopsis>Require client certificate (TLS ONLY)</synopsis>
825 <configOption name="type">
826 <synopsis>Must be of type 'transport'.</synopsis>
828 <configOption name="verify_client" default="false">
829 <synopsis>Require verification of client certificate (TLS ONLY)</synopsis>
831 <configOption name="verify_server" default="false">
832 <synopsis>Require verification of server certificate (TLS ONLY)</synopsis>
834 <configOption name="tos" default="false">
835 <synopsis>Enable TOS for the signalling sent over this transport</synopsis>
837 <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
838 for more information on this parameter.</para>
839 <note><para>This option does not apply to the <replaceable>ws</replaceable>
840 or the <replaceable>wss</replaceable> protocols.</para></note>
843 <configOption name="cos" default="false">
844 <synopsis>Enable COS for the signalling sent over this transport</synopsis>
846 <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
847 for more information on this parameter.</para>
848 <note><para>This option does not apply to the <replaceable>ws</replaceable>
849 or the <replaceable>wss</replaceable> protocols.</para></note>
853 <configObject name="contact">
854 <synopsis>A way of creating an aliased name to a SIP URI</synopsis>
856 Contacts are a way to hide SIP URIs from the dialplan directly.
857 They are also used to make a group of contactable parties when
858 in use with <literal>AoR</literal> lists.
859 </para></description>
860 <configOption name="type">
861 <synopsis>Must be of type 'contact'.</synopsis>
863 <configOption name="uri">
864 <synopsis>SIP URI to contact peer</synopsis>
866 <configOption name="expiration_time">
867 <synopsis>Time to keep alive a contact</synopsis>
869 Time to keep alive a contact. String style specification.
870 </para></description>
872 <configOption name="qualify_frequency" default="0">
873 <synopsis>Interval at which to qualify a contact</synopsis>
875 Interval between attempts to qualify the contact for reachability.
876 If <literal>0</literal> never qualify. Time in seconds.
877 </para></description>
879 <configOption name="outbound_proxy">
880 <synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
882 If set the provided URI will be used as the outbound proxy when an
883 OPTIONS request is sent to a contact for qualify purposes.
884 </para></description>
886 <configOption name="path">
887 <synopsis>Stored Path vector for use in Route headers on outgoing requests.</synopsis>
890 <configObject name="aor">
891 <synopsis>The configuration for a location of an endpoint</synopsis>
893 An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
894 AoRs are specified, an endpoint will not be reachable by Asterisk.
895 Beyond that, an AoR has other uses within Asterisk, such as inbound
898 An <literal>AoR</literal> is a way to allow dialing a group
899 of <literal>Contacts</literal> that all use the same
900 <literal>endpoint</literal> for calls.
902 This can be used as another way of grouping a list of contacts to dial
903 rather than specifing them each directly when dialing via the dialplan.
904 This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
906 Registrations: For Asterisk to match an inbound registration to an endpoint,
907 the AoR object name must match the user portion of the SIP URI in the "To:"
908 header of the inbound SIP registration. That will usually be equivalent
909 to the "user name" set in your hard or soft phones configuration.
910 </para></description>
911 <configOption name="contact">
912 <synopsis>Permanent contacts assigned to AoR</synopsis>
914 Contacts specified will be called whenever referenced
915 by <literal>chan_pjsip</literal>.
917 Use a separate "contact=" entry for each contact required. Contacts
918 are specified using a SIP URI.
919 </para></description>
921 <configOption name="default_expiration" default="3600">
922 <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
924 <configOption name="mailboxes">
925 <synopsis>Mailbox(es) to be associated with</synopsis>
926 <description><para>This option applies when an external entity subscribes to an AoR
927 for message waiting indications. The mailboxes specified will be subscribed to.
928 More than one mailbox can be specified with a comma-delimited string.</para></description>
930 <configOption name="maximum_expiration" default="7200">
931 <synopsis>Maximum time to keep an AoR</synopsis>
933 Maximium time to keep a peer with explicit expiration. Time in seconds.
934 </para></description>
936 <configOption name="max_contacts" default="0">
937 <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
939 Maximum number of contacts that can associate with this AoR. This value does
940 not affect the number of contacts that can be added with the "contact" option.
941 It only limits contacts added through external interaction, such as
944 <note><para>This should be set to <literal>1</literal> and
945 <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
946 wish to stick with the older <literal>chan_sip</literal> behaviour.
950 <configOption name="minimum_expiration" default="60">
951 <synopsis>Minimum keep alive time for an AoR</synopsis>
953 Minimum time to keep a peer with an explict expiration. Time in seconds.
954 </para></description>
956 <configOption name="remove_existing" default="no">
957 <synopsis>Determines whether new contacts replace existing ones.</synopsis>
959 On receiving a new registration to the AoR should it remove
960 the existing contact that was registered against it?
962 <note><para>This should be set to <literal>yes</literal> and
963 <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
964 wish to stick with the older <literal>chan_sip</literal> behaviour.
968 <configOption name="type">
969 <synopsis>Must be of type 'aor'.</synopsis>
971 <configOption name="qualify_frequency" default="0">
972 <synopsis>Interval at which to qualify an AoR</synopsis>
974 Interval between attempts to qualify the AoR for reachability.
975 If <literal>0</literal> never qualify. Time in seconds.
976 </para></description>
978 <configOption name="authenticate_qualify" default="no">
979 <synopsis>Authenticates a qualify request if needed</synopsis>
981 If true and a qualify request receives a challenge or authenticate response
982 authentication is attempted before declaring the contact available.
983 </para></description>
985 <configOption name="outbound_proxy">
986 <synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
988 If set the provided URI will be used as the outbound proxy when an
989 OPTIONS request is sent to a contact for qualify purposes.
990 </para></description>
992 <configOption name="support_path">
993 <synopsis>Enables Path support for REGISTER requests and Route support for other requests.</synopsis>
995 When this option is enabled, the Path headers in register requests will be saved
996 and its contents will be used in Route headers for outbound out-of-dialog requests
997 and in Path headers for outbound 200 responses. Path support will also be indicated
998 in the Supported header.
999 </para></description>
1002 <configObject name="system">
1003 <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
1005 The settings in this section are global. In addition to being global, the values will
1006 not be re-evaluated when a reload is performed. This is because the values must be set
1007 before the SIP stack is initialized. The only way to reset these values is to either
1008 restart Asterisk, or unload res_pjsip.so and then load it again.
1009 </para></description>
1010 <configOption name="timer_t1" default="500">
1011 <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
1013 Timer T1 is the base for determining how long to wait before retransmitting
1014 requests that receive no response when using an unreliable transport (e.g. UDP).
1015 For more information on this timer, see RFC 3261, Section 17.1.1.1.
1016 </para></description>
1018 <configOption name="timer_b" default="32000">
1019 <synopsis>Set transaction timer B value (milliseconds).</synopsis>
1021 Timer B determines the maximum amount of time to wait after sending an INVITE
1022 request before terminating the transaction. It is recommended that this be set
1023 to 64 * Timer T1, but it may be set higher if desired. For more information on
1024 this timer, see RFC 3261, Section 17.1.1.1.
1025 </para></description>
1027 <configOption name="compact_headers" default="no">
1028 <synopsis>Use the short forms of common SIP header names.</synopsis>
1030 <configOption name="threadpool_initial_size" default="0">
1031 <synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
1033 <configOption name="threadpool_auto_increment" default="5">
1034 <synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
1036 <configOption name="threadpool_idle_timeout" default="60">
1037 <synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
1039 <configOption name="threadpool_max_size" default="0">
1040 <synopsis>Maximum number of threads in the res_pjsip threadpool.
1041 A value of 0 indicates no maximum.</synopsis>
1043 <configOption name="type">
1044 <synopsis>Must be of type 'system'.</synopsis>
1047 <configObject name="global">
1048 <synopsis>Options that apply globally to all SIP communications</synopsis>
1050 The settings in this section are global. Unlike options in the <literal>system</literal>
1051 section, these options can be refreshed by performing a reload.
1052 </para></description>
1053 <configOption name="max_forwards" default="70">
1054 <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
1056 <configOption name="type">
1057 <synopsis>Must be of type 'global'.</synopsis>
1059 <configOption name="user_agent" default="Asterisk <Asterisk Version>">
1060 <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
1062 <configOption name="default_outbound_endpoint" default="default_outbound_endpoint">
1063 <synopsis>Endpoint to use when sending an outbound request to a URI without a specified endpoint.</synopsis>
1069 <manager name="PJSIPQualify" language="en_US">
1071 Qualify a chan_pjsip endpoint.
1074 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1075 <parameter name="Endpoint" required="true">
1076 <para>The endpoint you want to qualify.</para>
1080 <para>Qualify a chan_pjsip endpoint.</para>
1083 <manager name="PJSIPShowEndpoints" language="en_US">
1085 Lists PJSIP endpoints.
1090 Provides a listing of all endpoints. For each endpoint an <literal>EndpointList</literal> event
1091 is raised that contains relevant attributes and status information. Once all
1092 endpoints have been listed an <literal>EndpointListComplete</literal> event is issued.
1096 <manager name="PJSIPShowEndpoint" language="en_US">
1098 Detail listing of an endpoint and its objects.
1101 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1102 <parameter name="Endpoint" required="true">
1103 <para>The endpoint to list.</para>
1108 Provides a detailed listing of options for a given endpoint. Events are issued
1109 showing the configuration and status of the endpoint and associated objects. These
1110 events include <literal>EndpointDetail</literal>, <literal>AorDetail</literal>,
1111 <literal>AuthDetail</literal>, <literal>TransportDetail</literal>, and
1112 <literal>IdentifyDetail</literal>. Some events may be listed multiple times if multiple objects are
1113 associated (for instance AoRs). Once all detail events have been raised a final
1114 <literal>EndpointDetailComplete</literal> event is issued.
1120 #define MOD_DATA_CONTACT "contact"
1122 static pjsip_endpoint *ast_pjsip_endpoint;
1124 static struct ast_threadpool *sip_threadpool;
1126 static int register_service(void *data)
1128 pjsip_module **module = data;
1129 if (!ast_pjsip_endpoint) {
1130 ast_log(LOG_ERROR, "There is no PJSIP endpoint. Unable to register services\n");
1133 if (pjsip_endpt_register_module(ast_pjsip_endpoint, *module) != PJ_SUCCESS) {
1134 ast_log(LOG_ERROR, "Unable to register module %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1137 ast_debug(1, "Registered SIP service %.*s (%p)\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name), *module);
1138 ast_module_ref(ast_module_info->self);
1142 int ast_sip_register_service(pjsip_module *module)
1144 return ast_sip_push_task_synchronous(NULL, register_service, &module);
1147 static int unregister_service(void *data)
1149 pjsip_module **module = data;
1150 ast_module_unref(ast_module_info->self);
1151 if (!ast_pjsip_endpoint) {
1154 pjsip_endpt_unregister_module(ast_pjsip_endpoint, *module);
1155 ast_debug(1, "Unregistered SIP service %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1159 void ast_sip_unregister_service(pjsip_module *module)
1161 ast_sip_push_task_synchronous(NULL, unregister_service, &module);
1164 static struct ast_sip_authenticator *registered_authenticator;
1166 int ast_sip_register_authenticator(struct ast_sip_authenticator *auth)
1168 if (registered_authenticator) {
1169 ast_log(LOG_WARNING, "Authenticator %p is already registered. Cannot register a new one\n", registered_authenticator);
1172 registered_authenticator = auth;
1173 ast_debug(1, "Registered SIP authenticator module %p\n", auth);
1174 ast_module_ref(ast_module_info->self);
1178 void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth)
1180 if (registered_authenticator != auth) {
1181 ast_log(LOG_WARNING, "Trying to unregister authenticator %p but authenticator %p registered\n",
1182 auth, registered_authenticator);
1185 registered_authenticator = NULL;
1186 ast_debug(1, "Unregistered SIP authenticator %p\n", auth);
1187 ast_module_unref(ast_module_info->self);
1190 int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1192 if (!registered_authenticator) {
1193 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is not required\n");
1197 return registered_authenticator->requires_authentication(endpoint, rdata);
1200 enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
1201 pjsip_rx_data *rdata, pjsip_tx_data *tdata)
1203 if (!registered_authenticator) {
1204 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is successful\n");
1207 return registered_authenticator->check_authentication(endpoint, rdata, tdata);
1210 static struct ast_sip_outbound_authenticator *registered_outbound_authenticator;
1212 int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1214 if (registered_outbound_authenticator) {
1215 ast_log(LOG_WARNING, "Outbound authenticator %p is already registered. Cannot register a new one\n", registered_outbound_authenticator);
1218 registered_outbound_authenticator = auth;
1219 ast_debug(1, "Registered SIP outbound authenticator module %p\n", auth);
1220 ast_module_ref(ast_module_info->self);
1224 void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1226 if (registered_outbound_authenticator != auth) {
1227 ast_log(LOG_WARNING, "Trying to unregister outbound authenticator %p but outbound authenticator %p registered\n",
1228 auth, registered_outbound_authenticator);
1231 registered_outbound_authenticator = NULL;
1232 ast_debug(1, "Unregistered SIP outbound authenticator %p\n", auth);
1233 ast_module_unref(ast_module_info->self);
1236 int ast_sip_create_request_with_auth(const struct ast_sip_auth_vector *auths, pjsip_rx_data *challenge,
1237 pjsip_transaction *tsx, pjsip_tx_data **new_request)
1239 if (!registered_outbound_authenticator) {
1240 ast_log(LOG_WARNING, "No SIP outbound authenticator registered. Cannot respond to authentication challenge\n");
1243 return registered_outbound_authenticator->create_request_with_auth(auths, challenge, tsx, new_request);
1246 struct endpoint_identifier_list {
1247 struct ast_sip_endpoint_identifier *identifier;
1248 AST_RWLIST_ENTRY(endpoint_identifier_list) list;
1251 static AST_RWLIST_HEAD_STATIC(endpoint_identifiers, endpoint_identifier_list);
1253 int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1255 struct endpoint_identifier_list *id_list_item;
1256 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1258 id_list_item = ast_calloc(1, sizeof(*id_list_item));
1259 if (!id_list_item) {
1260 ast_log(LOG_ERROR, "Unabled to add endpoint identifier. Out of memory.\n");
1263 id_list_item->identifier = identifier;
1265 AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list);
1266 ast_debug(1, "Registered endpoint identifier %p\n", identifier);
1268 ast_module_ref(ast_module_info->self);
1272 void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1274 struct endpoint_identifier_list *iter;
1275 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1276 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) {
1277 if (iter->identifier == identifier) {
1278 AST_RWLIST_REMOVE_CURRENT(list);
1280 ast_debug(1, "Unregistered endpoint identifier %p\n", identifier);
1281 ast_module_unref(ast_module_info->self);
1285 AST_RWLIST_TRAVERSE_SAFE_END;
1288 struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata)
1290 struct endpoint_identifier_list *iter;
1291 struct ast_sip_endpoint *endpoint = NULL;
1292 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1293 AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) {
1294 ast_assert(iter->identifier->identify_endpoint != NULL);
1295 endpoint = iter->identifier->identify_endpoint(rdata);
1303 AST_RWLIST_HEAD_STATIC(endpoint_formatters, ast_sip_endpoint_formatter);
1305 int ast_sip_register_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
1307 SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1308 AST_RWLIST_INSERT_TAIL(&endpoint_formatters, obj, next);
1309 ast_module_ref(ast_module_info->self);
1313 void ast_sip_unregister_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
1315 struct ast_sip_endpoint_formatter *i;
1316 SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1317 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_formatters, i, next) {
1319 AST_RWLIST_REMOVE_CURRENT(next);
1320 ast_module_unref(ast_module_info->self);
1324 AST_RWLIST_TRAVERSE_SAFE_END;
1327 int ast_sip_format_endpoint_ami(struct ast_sip_endpoint *endpoint,
1328 struct ast_sip_ami *ami, int *count)
1331 struct ast_sip_endpoint_formatter *i;
1332 SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1334 AST_RWLIST_TRAVERSE(&endpoint_formatters, i, next) {
1335 if (i->format_ami && ((res = i->format_ami(endpoint, ami)) < 0)) {
1346 pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
1348 return ast_pjsip_endpoint;
1351 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)
1353 pj_str_t tmp, local_addr;
1355 pjsip_sip_uri *sip_uri;
1356 pjsip_transport_type_e type = PJSIP_TRANSPORT_UNSPECIFIED;
1358 char uuid_str[AST_UUID_STR_LEN];
1360 if (ast_strlen_zero(user)) {
1361 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1365 user = ast_uuid_to_str(uuid, uuid_str, sizeof(uuid_str));
1368 /* Parse the provided target URI so we can determine what transport it will end up using */
1369 pj_strdup_with_null(pool, &tmp, target);
1371 if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) ||
1372 (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
1376 sip_uri = pjsip_uri_get_uri(uri);
1378 /* Determine the transport type to use */
1379 if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
1380 type = PJSIP_TRANSPORT_TLS;
1381 } else if (!sip_uri->transport_param.slen) {
1382 type = PJSIP_TRANSPORT_UDP;
1384 type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
1387 if (type == PJSIP_TRANSPORT_UNSPECIFIED) {
1391 /* If the host is IPv6 turn the transport into an IPv6 version */
1392 if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1393 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1396 if (!ast_strlen_zero(domain)) {
1397 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1398 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1402 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1403 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1407 /* Get the local bound address for the transport that will be used when communicating with the provided URI */
1408 if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), pool, type, selector,
1409 &local_addr, &local_port) != PJ_SUCCESS) {
1411 /* If no local address can be retrieved using the transport manager use the host one */
1412 pj_strdup(pool, &local_addr, pj_gethostname());
1413 local_port = pjsip_transport_get_default_port_for_type(PJSIP_TRANSPORT_UDP);
1416 /* If IPv6 was specified in the transport, set the proper type */
1417 if (pj_strchr(&local_addr, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1418 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1421 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1422 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1423 "<sip:%s@%s%.*s%s:%d%s%s>",
1425 (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1426 (int)local_addr.slen,
1428 (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1430 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1431 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1436 static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
1438 RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
1439 const char *transport_name = endpoint->transport;
1441 if (ast_strlen_zero(transport_name)) {
1445 transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name);
1447 if (!transport || !transport->state) {
1448 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport '%s' for endpoint '%s'\n",
1449 transport_name, ast_sorcery_object_get_id(endpoint));
1453 if (transport->state->transport) {
1454 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1455 selector->u.transport = transport->state->transport;
1456 } else if (transport->state->factory) {
1457 selector->type = PJSIP_TPSELECTOR_LISTENER;
1458 selector->u.listener = transport->state->factory;
1466 pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
1468 char enclosed_uri[PJSIP_MAX_URL_SIZE];
1469 pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri, target_uri;
1470 pjsip_dialog *dlg = NULL;
1471 const char *outbound_proxy = endpoint->outbound_proxy;
1472 pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1473 static const pj_str_t HCONTACT = { "Contact", 7 };
1475 snprintf(enclosed_uri, sizeof(enclosed_uri), "<%s>", uri);
1476 pj_cstr(&remote_uri, enclosed_uri);
1478 pj_cstr(&target_uri, uri);
1480 if (pjsip_dlg_create_uac(pjsip_ua_instance(), &local_uri, NULL, &remote_uri, &target_uri, &dlg) != PJ_SUCCESS) {
1484 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1485 pjsip_dlg_terminate(dlg);
1489 if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
1490 pjsip_dlg_terminate(dlg);
1494 /* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
1495 pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
1496 dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
1497 dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
1499 /* If a request user has been specified and we are permitted to change it, do so */
1500 if (!ast_strlen_zero(request_user)) {
1501 pjsip_sip_uri *sip_uri;
1503 if (PJSIP_URI_SCHEME_IS_SIP(dlg->target) || PJSIP_URI_SCHEME_IS_SIPS(dlg->target)) {
1504 sip_uri = pjsip_uri_get_uri(dlg->target);
1505 pj_strdup2(dlg->pool, &sip_uri->user, request_user);
1507 if (PJSIP_URI_SCHEME_IS_SIP(dlg->remote.info->uri) || PJSIP_URI_SCHEME_IS_SIPS(dlg->remote.info->uri)) {
1508 sip_uri = pjsip_uri_get_uri(dlg->remote.info->uri);
1509 pj_strdup2(dlg->pool, &sip_uri->user, request_user);
1513 /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
1516 pjsip_dlg_set_transport(dlg, &selector);
1518 if (!ast_strlen_zero(outbound_proxy)) {
1519 pjsip_route_hdr route_set, *route;
1520 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1523 pj_list_init(&route_set);
1525 pj_strdup2_with_null(dlg->pool, &tmp, outbound_proxy);
1526 if (!(route = pjsip_parse_hdr(dlg->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1528 pjsip_dlg_terminate(dlg);
1531 pj_list_push_back(&route_set, route);
1533 pjsip_dlg_set_route_set(dlg, &route_set);
1541 pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1545 pjsip_transport_type_e type = rdata->tp_info.transport->key.type;
1548 contact.ptr = pj_pool_alloc(rdata->tp_info.pool, PJSIP_MAX_URL_SIZE);
1549 contact.slen = pj_ansi_snprintf(contact.ptr, PJSIP_MAX_URL_SIZE,
1550 "<sip:%s%.*s%s:%d%s%s>",
1551 (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1552 (int)rdata->tp_info.transport->local_name.host.slen,
1553 rdata->tp_info.transport->local_name.host.ptr,
1554 (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1555 rdata->tp_info.transport->local_name.port,
1556 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1557 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1559 status = pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, &contact, &dlg);
1560 if (status != PJ_SUCCESS) {
1561 char err[PJ_ERR_MSG_SIZE];
1563 pj_strerror(status, err, sizeof(err));
1564 ast_log(LOG_ERROR, "Could not create dialog with endpoint %s. %s\n",
1565 ast_sorcery_object_get_id(endpoint), err);
1572 /* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
1573 static const pjsip_method info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
1574 static const pjsip_method message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
1578 const pjsip_method *pmethod;
1580 { "INVITE", &pjsip_invite_method },
1581 { "CANCEL", &pjsip_cancel_method },
1582 { "ACK", &pjsip_ack_method },
1583 { "BYE", &pjsip_bye_method },
1584 { "REGISTER", &pjsip_register_method },
1585 { "OPTIONS", &pjsip_options_method },
1586 { "SUBSCRIBE", &pjsip_subscribe_method },
1587 { "NOTIFY", &pjsip_notify_method },
1588 { "PUBLISH", &pjsip_publish_method },
1589 { "INFO", &info_method },
1590 { "MESSAGE", &message_method },
1593 static const pjsip_method *get_pjsip_method(const char *method)
1596 for (i = 0; i < ARRAY_LEN(methods); ++i) {
1597 if (!strcmp(method, methods[i].method)) {
1598 return methods[i].pmethod;
1604 static int create_in_dialog_request(const pjsip_method *method, struct pjsip_dialog *dlg, pjsip_tx_data **tdata)
1606 if (pjsip_dlg_create_request(dlg, method, -1, tdata) != PJ_SUCCESS) {
1607 ast_log(LOG_WARNING, "Unable to create in-dialog request.\n");
1614 static pj_bool_t supplement_on_rx_request(pjsip_rx_data *rdata);
1615 static pjsip_module supplement_module = {
1616 .name = { "Out of dialog supplement hook", 29 },
1618 .priority = PJSIP_MOD_PRIORITY_APPLICATION - 1,
1619 .on_rx_request = supplement_on_rx_request,
1622 static int create_out_of_dialog_request(const pjsip_method *method, struct ast_sip_endpoint *endpoint,
1623 const char *uri, struct ast_sip_contact *provided_contact, pjsip_tx_data **tdata)
1625 RAII_VAR(struct ast_sip_contact *, contact, ao2_bump(provided_contact), ao2_cleanup);
1626 pj_str_t remote_uri;
1629 pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1631 if (ast_strlen_zero(uri)) {
1632 if (!endpoint && !contact) {
1633 ast_log(LOG_ERROR, "An endpoint and/or uri must be specified\n");
1638 contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
1640 if (!contact || ast_strlen_zero(contact->uri)) {
1641 ast_log(LOG_ERROR, "Unable to retrieve contact for endpoint %s\n",
1642 ast_sorcery_object_get_id(endpoint));
1646 pj_cstr(&remote_uri, contact->uri);
1648 pj_cstr(&remote_uri, uri);
1652 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1653 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
1654 ast_sorcery_object_get_id(endpoint));
1659 pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
1662 ast_log(LOG_ERROR, "Unable to create PJLIB memory pool\n");
1666 if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL,
1667 endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
1668 ast_log(LOG_ERROR, "Unable to create From header for %.*s request to endpoint %s\n",
1669 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1670 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1674 if (pjsip_endpt_create_request(ast_sip_get_pjsip_endpoint(), method, &remote_uri,
1675 &from, &remote_uri, &from, NULL, -1, NULL, tdata) != PJ_SUCCESS) {
1676 ast_log(LOG_ERROR, "Unable to create outbound %.*s request to endpoint %s\n",
1677 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1678 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1682 /* If an outbound proxy is specified on the endpoint apply it to this request */
1683 if (endpoint && !ast_strlen_zero(endpoint->outbound_proxy) &&
1684 ast_sip_set_outbound_proxy((*tdata), endpoint->outbound_proxy)) {
1685 ast_log(LOG_ERROR, "Unable to apply outbound proxy on request %.*s to endpoint %s\n",
1686 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1687 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1691 ast_sip_mod_data_set((*tdata)->pool, (*tdata)->mod_data, supplement_module.id, MOD_DATA_CONTACT, ao2_bump(contact));
1693 /* We can release this pool since request creation copied all the necessary
1694 * data into the outbound request's pool
1696 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1700 int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
1701 struct ast_sip_endpoint *endpoint, const char *uri,
1702 struct ast_sip_contact *contact, pjsip_tx_data **tdata)
1704 const pjsip_method *pmethod = get_pjsip_method(method);
1707 ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
1712 return create_in_dialog_request(pmethod, dlg, tdata);
1714 return create_out_of_dialog_request(pmethod, endpoint, uri, contact, tdata);
1718 AST_RWLIST_HEAD_STATIC(supplements, ast_sip_supplement);
1720 int ast_sip_register_supplement(struct ast_sip_supplement *supplement)
1722 struct ast_sip_supplement *iter;
1724 SCOPED_LOCK(lock, &supplements, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1726 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&supplements, iter, next) {
1727 if (iter->priority > supplement->priority) {
1728 AST_RWLIST_INSERT_BEFORE_CURRENT(supplement, next);
1733 AST_RWLIST_TRAVERSE_SAFE_END;
1736 AST_RWLIST_INSERT_TAIL(&supplements, supplement, next);
1738 ast_module_ref(ast_module_info->self);
1742 void ast_sip_unregister_supplement(struct ast_sip_supplement *supplement)
1744 struct ast_sip_supplement *iter;
1745 SCOPED_LOCK(lock, &supplements, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1746 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&supplements, iter, next) {
1747 if (supplement == iter) {
1748 AST_RWLIST_REMOVE_CURRENT(next);
1749 ast_module_unref(ast_module_info->self);
1753 AST_RWLIST_TRAVERSE_SAFE_END;
1756 static int send_in_dialog_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg)
1758 if (pjsip_dlg_send_request(dlg, tdata, -1, NULL) != PJ_SUCCESS) {
1759 ast_log(LOG_WARNING, "Unable to send in-dialog request.\n");
1765 static pj_bool_t does_method_match(const pj_str_t *message_method, const char *supplement_method)
1769 if (ast_strlen_zero(supplement_method)) {
1773 pj_cstr(&method, supplement_method);
1775 return pj_stristr(&method, message_method) ? PJ_TRUE : PJ_FALSE;
1778 /*! \brief Structure to hold information about an outbound request */
1779 struct send_request_data {
1780 struct ast_sip_endpoint *endpoint; /*! The endpoint associated with this request */
1781 void *token; /*! Information to be provided to the callback upon receipt of a response */
1782 void (*callback)(void *token, pjsip_event *e); /*! The callback to be called upon receipt of a response */
1785 static void send_request_data_destroy(void *obj)
1787 struct send_request_data *req_data = obj;
1788 ao2_cleanup(req_data->endpoint);
1791 static struct send_request_data *send_request_data_alloc(struct ast_sip_endpoint *endpoint,
1792 void *token, void (*callback)(void *token, pjsip_event *e))
1794 struct send_request_data *req_data = ao2_alloc(sizeof(*req_data), send_request_data_destroy);
1800 req_data->endpoint = ao2_bump(endpoint);
1801 req_data->token = token;
1802 req_data->callback = callback;
1807 static void send_request_cb(void *token, pjsip_event *e)
1809 RAII_VAR(struct send_request_data *, req_data, token, ao2_cleanup);
1810 pjsip_transaction *tsx = e->body.tsx_state.tsx;
1811 pjsip_rx_data *challenge = e->body.tsx_state.src.rdata;
1812 pjsip_tx_data *tdata;
1813 struct ast_sip_supplement *supplement;
1815 AST_RWLIST_RDLOCK(&supplements);
1816 AST_LIST_TRAVERSE(&supplements, supplement, next) {
1817 if (supplement->incoming_response && does_method_match(&challenge->msg_info.cseq->method.name, supplement->method)) {
1818 supplement->incoming_response(req_data->endpoint, challenge);
1821 AST_RWLIST_UNLOCK(&supplements);
1823 if (tsx->status_code == 401 || tsx->status_code == 407) {
1824 if (!ast_sip_create_request_with_auth(&req_data->endpoint->outbound_auths, challenge, tsx, &tdata)) {
1825 pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, req_data->token, req_data->callback);
1830 if (req_data->callback) {
1831 req_data->callback(req_data->token, e);
1835 static int send_out_of_dialog_request(pjsip_tx_data *tdata, struct ast_sip_endpoint *endpoint,
1836 void *token, void (*callback)(void *token, pjsip_event *e))
1838 struct ast_sip_supplement *supplement;
1839 struct send_request_data *req_data = send_request_data_alloc(endpoint, token, callback);
1840 struct ast_sip_contact *contact = ast_sip_mod_data_get(tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT);
1846 AST_RWLIST_RDLOCK(&supplements);
1847 AST_LIST_TRAVERSE(&supplements, supplement, next) {
1848 if (supplement->outgoing_request && does_method_match(&tdata->msg->line.req.method.name, supplement->method)) {
1849 supplement->outgoing_request(endpoint, contact, tdata);
1852 AST_RWLIST_UNLOCK(&supplements);
1854 ast_sip_mod_data_set(tdata->pool, tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT, NULL);
1855 ao2_cleanup(contact);
1857 if (pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, req_data, send_request_cb) != PJ_SUCCESS) {
1858 ast_log(LOG_ERROR, "Error attempting to send outbound %.*s request to endpoint %s\n",
1859 (int) pj_strlen(&tdata->msg->line.req.method.name),
1860 pj_strbuf(&tdata->msg->line.req.method.name),
1861 ast_sorcery_object_get_id(endpoint));
1862 ao2_cleanup(req_data);
1869 int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg,
1870 struct ast_sip_endpoint *endpoint, void *token,
1871 void (*callback)(void *token, pjsip_event *e))
1873 ast_assert(tdata->msg->type == PJSIP_REQUEST_MSG);
1876 return send_in_dialog_request(tdata, dlg);
1878 return send_out_of_dialog_request(tdata, endpoint, token, callback);
1882 int ast_sip_set_outbound_proxy(pjsip_tx_data *tdata, const char *proxy)
1884 pjsip_route_hdr *route;
1885 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1888 pj_strdup2_with_null(tdata->pool, &tmp, proxy);
1889 if (!(route = pjsip_parse_hdr(tdata->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1893 pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr*)route);
1898 int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value)
1902 pjsip_generic_string_hdr *hdr;
1904 pj_cstr(&hdr_name, name);
1905 pj_cstr(&hdr_value, value);
1907 hdr = pjsip_generic_string_hdr_create(tdata->pool, &hdr_name, &hdr_value);
1909 pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) hdr);
1913 static pjsip_msg_body *ast_body_to_pjsip_body(pj_pool_t *pool, const struct ast_sip_body *body)
1919 pj_cstr(&type, body->type);
1920 pj_cstr(&subtype, body->subtype);
1921 pj_cstr(&body_text, body->body_text);
1923 return pjsip_msg_body_create(pool, &type, &subtype, &body_text);
1926 int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body)
1928 pjsip_msg_body *pjsip_body = ast_body_to_pjsip_body(tdata->pool, body);
1929 tdata->msg->body = pjsip_body;
1933 int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies)
1936 /* NULL for type and subtype automatically creates "multipart/mixed" */
1937 pjsip_msg_body *body = pjsip_multipart_create(tdata->pool, NULL, NULL);
1939 for (i = 0; i < num_bodies; ++i) {
1940 pjsip_multipart_part *part = pjsip_multipart_create_part(tdata->pool);
1941 part->body = ast_body_to_pjsip_body(tdata->pool, bodies[i]);
1942 pjsip_multipart_add_part(tdata->pool, body, part);
1945 tdata->msg->body = body;
1949 int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text)
1951 size_t combined_size = strlen(body_text) + tdata->msg->body->len;
1952 struct ast_str *body_buffer = ast_str_alloca(combined_size);
1954 ast_str_set(&body_buffer, 0, "%.*s%s", (int) tdata->msg->body->len, (char *) tdata->msg->body->data, body_text);
1956 tdata->msg->body->data = pj_pool_alloc(tdata->pool, combined_size);
1957 pj_memcpy(tdata->msg->body->data, ast_str_buffer(body_buffer), combined_size);
1958 tdata->msg->body->len = combined_size;
1963 struct ast_taskprocessor *ast_sip_create_serializer(void)
1965 struct ast_taskprocessor *serializer;
1966 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1967 char name[AST_UUID_STR_LEN];
1973 ast_uuid_to_str(uuid, name, sizeof(name));
1975 serializer = ast_threadpool_serializer(name, sip_threadpool);
1982 int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1985 return ast_taskprocessor_push(serializer, sip_task, task_data);
1987 return ast_threadpool_push(sip_threadpool, sip_task, task_data);
1991 struct sync_task_data {
1996 int (*task)(void *);
2000 static int sync_task(void *data)
2002 struct sync_task_data *std = data;
2003 std->fail = std->task(std->task_data);
2005 ast_mutex_lock(&std->lock);
2007 ast_cond_signal(&std->cond);
2008 ast_mutex_unlock(&std->lock);
2012 int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
2014 /* This method is an onion */
2015 struct sync_task_data std;
2017 if (ast_sip_thread_is_servant()) {
2018 return sip_task(task_data);
2021 ast_mutex_init(&std.lock);
2022 ast_cond_init(&std.cond, NULL);
2023 std.fail = std.complete = 0;
2024 std.task = sip_task;
2025 std.task_data = task_data;
2028 if (ast_taskprocessor_push(serializer, sync_task, &std)) {
2032 if (ast_threadpool_push(sip_threadpool, sync_task, &std)) {
2037 ast_mutex_lock(&std.lock);
2038 while (!std.complete) {
2039 ast_cond_wait(&std.cond, &std.lock);
2041 ast_mutex_unlock(&std.lock);
2043 ast_mutex_destroy(&std.lock);
2044 ast_cond_destroy(&std.cond);
2048 void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
2050 size_t chars_to_copy = MIN(size - 1, pj_strlen(src));
2051 memcpy(dest, pj_strbuf(src), chars_to_copy);
2052 dest[chars_to_copy] = '\0';
2055 int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype)
2057 pjsip_media_type compare;
2059 if (!content_type) {
2063 pjsip_media_type_init2(&compare, type, subtype);
2065 return pjsip_media_type_cmp(content_type, &compare, 0) ? 0 : -1;
2068 pj_caching_pool caching_pool;
2069 pj_pool_t *memory_pool;
2070 pj_thread_t *monitor_thread;
2071 static int monitor_continue;
2073 static void *monitor_thread_exec(void *endpt)
2075 while (monitor_continue) {
2076 const pj_time_val delay = {0, 10};
2077 pjsip_endpt_handle_events(ast_pjsip_endpoint, &delay);
2082 static void stop_monitor_thread(void)
2084 monitor_continue = 0;
2085 pj_thread_join(monitor_thread);
2088 AST_THREADSTORAGE(pj_thread_storage);
2089 AST_THREADSTORAGE(servant_id_storage);
2090 #define SIP_SERVANT_ID 0x5E2F1D
2092 static void sip_thread_start(void)
2094 pj_thread_desc *desc;
2095 pj_thread_t *thread;
2096 uint32_t *servant_id;
2098 servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
2100 ast_log(LOG_ERROR, "Could not set SIP servant ID in thread-local storage.\n");
2103 *servant_id = SIP_SERVANT_ID;
2105 desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc));
2107 ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n");
2110 pj_bzero(*desc, sizeof(*desc));
2112 if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) {
2113 ast_log(LOG_ERROR, "Couldn't register thread with PJLIB.\n");
2117 int ast_sip_thread_is_servant(void)
2119 uint32_t *servant_id;
2121 servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
2126 return *servant_id == SIP_SERVANT_ID;
2129 void *ast_sip_dict_get(void *ht, const char *key)
2131 unsigned int hval = 0;
2137 return pj_hash_get(ht, key, PJ_HASH_KEY_STRING, &hval);
2140 void *ast_sip_dict_set(pj_pool_t* pool, void *ht,
2141 const char *key, void *val)
2144 ht = pj_hash_create(pool, 11);
2147 pj_hash_set(pool, ht, key, PJ_HASH_KEY_STRING, 0, val);
2152 static pj_bool_t supplement_on_rx_request(pjsip_rx_data *rdata)
2154 struct ast_sip_supplement *supplement;
2156 if (pjsip_rdata_get_dlg(rdata)) {
2160 AST_RWLIST_RDLOCK(&supplements);
2161 AST_LIST_TRAVERSE(&supplements, supplement, next) {
2162 if (supplement->incoming_request && does_method_match(&rdata->msg_info.msg->line.req.method.name, supplement->method)) {
2163 supplement->incoming_request(ast_pjsip_rdata_get_endpoint(rdata), rdata);
2166 AST_RWLIST_UNLOCK(&supplements);
2171 int ast_sip_send_response(pjsip_response_addr *res_addr, pjsip_tx_data *tdata, struct ast_sip_endpoint *sip_endpoint)
2173 struct ast_sip_supplement *supplement;
2174 pjsip_cseq_hdr *cseq = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_CSEQ, NULL);
2175 struct ast_sip_contact *contact = ast_sip_mod_data_get(tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT);
2177 AST_RWLIST_RDLOCK(&supplements);
2178 AST_LIST_TRAVERSE(&supplements, supplement, next) {
2179 if (supplement->outgoing_response && does_method_match(&cseq->method.name, supplement->method)) {
2180 supplement->outgoing_response(sip_endpoint, contact, tdata);
2183 AST_RWLIST_UNLOCK(&supplements);
2185 ast_sip_mod_data_set(tdata->pool, tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT, NULL);
2186 ao2_cleanup(contact);
2188 return pjsip_endpt_send_response(ast_sip_get_pjsip_endpoint(), res_addr, tdata, NULL, NULL);
2191 int ast_sip_create_response(const pjsip_rx_data *rdata, int st_code,
2192 struct ast_sip_contact *contact, pjsip_tx_data **tdata)
2194 int res = pjsip_endpt_create_response(ast_sip_get_pjsip_endpoint(), rdata, st_code, NULL, tdata);
2197 ast_sip_mod_data_set((*tdata)->pool, (*tdata)->mod_data, supplement_module.id, MOD_DATA_CONTACT, ao2_bump(contact));
2203 static void remove_request_headers(pjsip_endpoint *endpt)
2205 const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt);
2206 pjsip_hdr *iter = request_headers->next;
2208 while (iter != request_headers) {
2209 pjsip_hdr *to_erase = iter;
2211 pj_list_erase(to_erase);
2215 static int load_module(void)
2217 /* The third parameter is just copied from
2218 * example code from PJLIB. This can be adjusted
2222 struct ast_threadpool_options options;
2224 if (pj_init() != PJ_SUCCESS) {
2225 return AST_MODULE_LOAD_DECLINE;
2228 if (pjlib_util_init() != PJ_SUCCESS) {
2230 return AST_MODULE_LOAD_DECLINE;
2233 pj_caching_pool_init(&caching_pool, NULL, 1024 * 1024);
2234 if (pjsip_endpt_create(&caching_pool.factory, "SIP", &ast_pjsip_endpoint) != PJ_SUCCESS) {
2235 ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n");
2236 pj_caching_pool_destroy(&caching_pool);
2237 return AST_MODULE_LOAD_DECLINE;
2240 /* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that,
2241 * we need to stop PJSIP from doing it automatically
2243 remove_request_headers(ast_pjsip_endpoint);
2245 memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL);
2247 ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n");
2248 pjsip_endpt_destroy(ast_pjsip_endpoint);
2249 ast_pjsip_endpoint = NULL;
2250 pj_caching_pool_destroy(&caching_pool);
2251 return AST_MODULE_LOAD_DECLINE;
2254 if (ast_sip_initialize_system()) {
2255 ast_log(LOG_ERROR, "Failed to initialize SIP 'system' configuration section. Aborting load\n");
2256 pj_pool_release(memory_pool);
2258 pjsip_endpt_destroy(ast_pjsip_endpoint);
2259 ast_pjsip_endpoint = NULL;
2260 pj_caching_pool_destroy(&caching_pool);
2261 return AST_MODULE_LOAD_DECLINE;
2264 sip_get_threadpool_options(&options);
2265 options.thread_start = sip_thread_start;
2266 sip_threadpool = ast_threadpool_create("SIP", NULL, &options);
2267 if (!sip_threadpool) {
2268 ast_log(LOG_ERROR, "Failed to create SIP threadpool. Aborting load\n");
2269 pj_pool_release(memory_pool);
2271 pjsip_endpt_destroy(ast_pjsip_endpoint);
2272 ast_pjsip_endpoint = NULL;
2273 pj_caching_pool_destroy(&caching_pool);
2274 return AST_MODULE_LOAD_DECLINE;
2277 pjsip_tsx_layer_init_module(ast_pjsip_endpoint);
2278 pjsip_ua_init_module(ast_pjsip_endpoint, NULL);
2280 monitor_continue = 1;
2281 status = pj_thread_create(memory_pool, "SIP", (pj_thread_proc *) &monitor_thread_exec,
2282 NULL, PJ_THREAD_DEFAULT_STACK_SIZE * 2, 0, &monitor_thread);
2283 if (status != PJ_SUCCESS) {
2284 ast_log(LOG_ERROR, "Failed to start SIP monitor thread. 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 ast_sip_initialize_global_headers();
2295 if (ast_res_pjsip_initialize_configuration(ast_module_info)) {
2296 ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
2297 ast_sip_destroy_global_headers();
2298 stop_monitor_thread();
2299 pj_pool_release(memory_pool);
2301 pjsip_endpt_destroy(ast_pjsip_endpoint);
2302 ast_pjsip_endpoint = NULL;
2303 pj_caching_pool_destroy(&caching_pool);
2304 return AST_MODULE_LOAD_DECLINE;
2307 if (ast_sip_initialize_distributor()) {
2308 ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n");
2309 ast_res_pjsip_destroy_configuration();
2310 ast_sip_destroy_global_headers();
2311 stop_monitor_thread();
2312 pj_pool_release(memory_pool);
2314 pjsip_endpt_destroy(ast_pjsip_endpoint);
2315 ast_pjsip_endpoint = NULL;
2316 pj_caching_pool_destroy(&caching_pool);
2317 return AST_MODULE_LOAD_DECLINE;
2320 if (ast_sip_register_service(&supplement_module)) {
2321 ast_log(LOG_ERROR, "Failed to initialize supplement hooks. Aborting load\n");
2322 ast_sip_destroy_distributor();
2323 ast_res_pjsip_destroy_configuration();
2324 ast_sip_destroy_global_headers();
2325 stop_monitor_thread();
2326 pj_pool_release(memory_pool);
2328 pjsip_endpt_destroy(ast_pjsip_endpoint);
2329 ast_pjsip_endpoint = NULL;
2330 pj_caching_pool_destroy(&caching_pool);
2331 return AST_MODULE_LOAD_DECLINE;
2334 if (ast_sip_initialize_outbound_authentication()) {
2335 ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n");
2336 ast_sip_unregister_service(&supplement_module);
2337 ast_sip_destroy_distributor();
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 ast_res_pjsip_init_options_handling(0);
2351 ast_module_ref(ast_module_info->self);
2353 return AST_MODULE_LOAD_SUCCESS;
2356 static int reload_module(void)
2358 if (ast_res_pjsip_reload_configuration()) {
2359 return AST_MODULE_LOAD_DECLINE;
2361 ast_res_pjsip_init_options_handling(1);
2365 static int unload_module(void)
2367 /* This will never get called as this module can't be unloaded */
2371 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Basic SIP resource",
2372 .load = load_module,
2373 .unload = unload_module,
2374 .reload = reload_module,
2375 .load_pri = AST_MODPRI_CHANNEL_DEPEND - 5,
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