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 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 the AVP or SAVP RTP
388 profile for all media offers on outbound calls and media updates, but will
389 accept either the AVP/AVPF or SAVP/SAVPF RTP profile for all inbound
391 </para></description>
393 <configOption name="media_encryption" default="no">
394 <synopsis>Determines whether res_pjsip will use and enforce usage of media encryption
395 for this endpoint.</synopsis>
398 <enum name="no"><para>
399 res_pjsip will offer no encryption and allow no encryption to be setup.
401 <enum name="sdes"><para>
402 res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP
403 transport should be used in conjunction with this option to prevent
404 exposure of media encryption keys.
406 <enum name="dtls"><para>
407 res_pjsip will offer DTLS-SRTP setup.
412 <configOption name="inband_progress" default="no">
413 <synopsis>Determines whether chan_pjsip will indicate ringing using inband
416 If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
417 when told to indicate ringing and will immediately start sending ringing
420 If set to <literal>no</literal>, chan_pjsip will send a 180 Ringing when told
421 to indicate ringing and will NOT send it as audio.
422 </para></description>
424 <configOption name="call_group">
425 <synopsis>The numeric pickup groups for a channel.</synopsis>
427 Can be set to a comma separated list of numbers or ranges between the values
428 of 0-63 (maximum of 64 groups).
429 </para></description>
431 <configOption name="pickup_group">
432 <synopsis>The numeric pickup groups that a channel can pickup.</synopsis>
434 Can be set to a comma separated list of numbers or ranges between the values
435 of 0-63 (maximum of 64 groups).
436 </para></description>
438 <configOption name="named_call_group">
439 <synopsis>The named pickup groups for a channel.</synopsis>
441 Can be set to a comma separated list of case sensitive strings limited by
442 supported line length.
443 </para></description>
445 <configOption name="named_pickup_group">
446 <synopsis>The named pickup groups that a channel can pickup.</synopsis>
448 Can be set to a comma separated list of case sensitive strings limited by
449 supported line length.
450 </para></description>
452 <configOption name="device_state_busy_at" default="0">
453 <synopsis>The number of in-use channels which will cause busy to be returned as device state</synopsis>
455 When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the
456 PJSIP channel driver will return busy as the device state instead of in use.
457 </para></description>
459 <configOption name="t38_udptl" default="no">
460 <synopsis>Whether T.38 UDPTL support is enabled or not</synopsis>
462 If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted
464 </para></description>
466 <configOption name="t38_udptl_ec" default="none">
467 <synopsis>T.38 UDPTL error correction method</synopsis>
470 <enum name="none"><para>
471 No error correction should be used.
473 <enum name="fec"><para>
474 Forward error correction should be used.
476 <enum name="redundancy"><para>
477 Redundacy error correction should be used.
482 <configOption name="t38_udptl_maxdatagram" default="0">
483 <synopsis>T.38 UDPTL maximum datagram size</synopsis>
485 This option can be set to override the maximum datagram of a remote endpoint for broken
487 </para></description>
489 <configOption name="fax_detect" default="no">
490 <synopsis>Whether CNG tone detection is enabled</synopsis>
492 This option can be set to send the session to the fax extension when a CNG tone is
494 </para></description>
496 <configOption name="t38_udptl_nat" default="no">
497 <synopsis>Whether NAT support is enabled on UDPTL sessions</synopsis>
499 When enabled the UDPTL stack will send UDPTL packets to the source address of
501 </para></description>
503 <configOption name="t38_udptl_ipv6" default="no">
504 <synopsis>Whether IPv6 is used for UDPTL Sessions</synopsis>
506 When enabled the UDPTL stack will use IPv6.
507 </para></description>
509 <configOption name="tone_zone">
510 <synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis>
512 <configOption name="language">
513 <synopsis>Set the default language to use for channels created for this endpoint.</synopsis>
515 <configOption name="one_touch_recording" default="no">
516 <synopsis>Determines whether one-touch recording is allowed for this endpoint.</synopsis>
518 <ref type="configOption">recordonfeature</ref>
519 <ref type="configOption">recordofffeature</ref>
522 <configOption name="record_on_feature" default="automixmon">
523 <synopsis>The feature to enact when one-touch recording is turned on.</synopsis>
525 <para>When an INFO request for one-touch recording arrives with a Record header set to "on", this
526 feature will be enabled for the channel. The feature designated here can be any built-in
527 or dynamic feature defined in features.conf.</para>
528 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
531 <ref type="configOption">one_touch_recording</ref>
532 <ref type="configOption">recordofffeature</ref>
535 <configOption name="record_off_feature" default="automixmon">
536 <synopsis>The feature to enact when one-touch recording is turned off.</synopsis>
538 <para>When an INFO request for one-touch recording arrives with a Record header set to "off", this
539 feature will be enabled for the channel. The feature designated here can be any built-in
540 or dynamic feature defined in features.conf.</para>
541 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
544 <ref type="configOption">one_touch_recording</ref>
545 <ref type="configOption">recordonfeature</ref>
548 <configOption name="rtp_engine" default="asterisk">
549 <synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis>
551 <configOption name="allow_transfer" default="yes">
552 <synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis>
554 <configOption name="sdp_owner" default="-">
555 <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
557 <configOption name="sdp_session" default="Asterisk">
558 <synopsis>String used for the SDP session (s=) line.</synopsis>
560 <configOption name="tos_audio">
561 <synopsis>DSCP TOS bits 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="tos_video">
567 <synopsis>DSCP TOS bits 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="cos_audio">
573 <synopsis>Priority for audio streams</synopsis>
575 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
576 </para></description>
578 <configOption name="cos_video">
579 <synopsis>Priority for video streams</synopsis>
581 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
582 </para></description>
584 <configOption name="allow_subscribe" default="yes">
585 <synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis>
587 <configOption name="sub_min_expiry" default="60">
588 <synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis>
590 <configOption name="from_user">
591 <synopsis>Username to use in From header for requests to this endpoint.</synopsis>
593 <configOption name="mwi_from_user">
594 <synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis>
596 <configOption name="from_domain">
597 <synopsis>Domain to user in From header for requests to this endpoint.</synopsis>
599 <configOption name="dtls_verify">
600 <synopsis>Verify that the provided peer certificate is valid</synopsis>
602 This option only applies if <replaceable>media_encryption</replaceable> is
603 set to <literal>dtls</literal>.
604 </para></description>
606 <configOption name="dtls_rekey">
607 <synopsis>Interval at which to renegotiate the TLS session and rekey the SRTP session</synopsis>
609 This option only applies if <replaceable>media_encryption</replaceable> is
610 set to <literal>dtls</literal>.
612 If this is not set or the value provided is 0 rekeying will be disabled.
613 </para></description>
615 <configOption name="dtls_cert_file">
616 <synopsis>Path to certificate file to present to peer</synopsis>
618 This option only applies if <replaceable>media_encryption</replaceable> is
619 set to <literal>dtls</literal>.
620 </para></description>
622 <configOption name="dtls_private_key">
623 <synopsis>Path to private key for certificate file</synopsis>
625 This option only applies if <replaceable>media_encryption</replaceable> is
626 set to <literal>dtls</literal>.
627 </para></description>
629 <configOption name="dtls_cipher">
630 <synopsis>Cipher to use for DTLS negotiation</synopsis>
632 This option only applies if <replaceable>media_encryption</replaceable> is
633 set to <literal>dtls</literal>.
635 Many options for acceptable ciphers. See link for more:
636 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
637 </para></description>
639 <configOption name="dtls_ca_file">
640 <synopsis>Path to certificate authority certificate</synopsis>
642 This option only applies if <replaceable>media_encryption</replaceable> is
643 set to <literal>dtls</literal>.
644 </para></description>
646 <configOption name="dtls_ca_path">
647 <synopsis>Path to a directory containing certificate authority certificates</synopsis>
649 This option only applies if <replaceable>media_encryption</replaceable> is
650 set to <literal>dtls</literal>.
651 </para></description>
653 <configOption name="dtls_setup">
654 <synopsis>Whether we are willing to accept connections, connect to the other party, or both.</synopsis>
657 This option only applies if <replaceable>media_encryption</replaceable> is
658 set to <literal>dtls</literal>.
661 <enum name="active"><para>
662 res_pjsip will make a connection to the peer.
664 <enum name="passive"><para>
665 res_pjsip will accept connections from the peer.
667 <enum name="actpass"><para>
668 res_pjsip will offer and accept connections from the peer.
673 <configOption name="srtp_tag_32">
674 <synopsis>Determines whether 32 byte tags should be used instead of 80 byte tags.</synopsis>
676 This option only applies if <replaceable>media_encryption</replaceable> is
677 set to <literal>sdes</literal> or <literal>dtls</literal>.
678 </para></description>
680 <configOption name="set_var">
681 <synopsis>Variable set on a channel involving the endpoint.</synopsis>
683 When a new channel is created using the endpoint set the specified
684 variable(s) on that channel. For multiple channel variables specify
685 multiple 'set_var'(s).
686 </para></description>
689 <configObject name="auth">
690 <synopsis>Authentication type</synopsis>
692 Authentication objects hold the authentication information for use
693 by other objects such as <literal>endpoints</literal> or <literal>registrations</literal>.
694 This also allows for multiple objects to use a single auth object. See
695 the <literal>auth_type</literal> config option for password style choices.
696 </para></description>
697 <configOption name="auth_type" default="userpass">
698 <synopsis>Authentication type</synopsis>
700 This option specifies which of the password style config options should be read
701 when trying to authenticate an endpoint inbound request. If set to <literal>userpass</literal>
702 then we'll read from the 'password' option. For <literal>md5</literal> we'll read
707 <enum name="userpass"/>
711 <configOption name="nonce_lifetime" default="32">
712 <synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
714 <configOption name="md5_cred">
715 <synopsis>MD5 Hash used for authentication.</synopsis>
716 <description><para>Only used when auth_type is <literal>md5</literal>.</para></description>
718 <configOption name="password">
719 <synopsis>PlainText password used for authentication.</synopsis>
720 <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
722 <configOption name="realm" default="asterisk">
723 <synopsis>SIP realm for endpoint</synopsis>
725 <configOption name="type">
726 <synopsis>Must be 'auth'</synopsis>
728 <configOption name="username">
729 <synopsis>Username to use for account</synopsis>
732 <configObject name="domain_alias">
733 <synopsis>Domain Alias</synopsis>
735 Signifies that a domain is an alias. If the domain on a session is
736 not found to match an AoR then this object is used to see if we have
737 an alias for the AoR to which the endpoint is binding. This objects
738 name as defined in configuration should be the domain alias and a
739 config option is provided to specify the domain to be aliased.
740 </para></description>
741 <configOption name="type">
742 <synopsis>Must be of type 'domain_alias'.</synopsis>
744 <configOption name="domain">
745 <synopsis>Domain to be aliased</synopsis>
748 <configObject name="transport">
749 <synopsis>SIP Transport</synopsis>
751 <emphasis>Transports</emphasis>
753 <para>There are different transports and protocol derivatives
754 supported by <literal>res_pjsip</literal>. They are in order of
755 preference: UDP, TCP, and WebSocket (WS).</para>
756 <note><para>Changes to transport configuration in pjsip.conf will only be
757 effected on a complete restart of Asterisk. A module reload
758 will not suffice.</para></note>
760 <configOption name="async_operations" default="1">
761 <synopsis>Number of simultaneous Asynchronous Operations</synopsis>
763 <configOption name="bind">
764 <synopsis>IP Address and optional port to bind to for this transport</synopsis>
766 <configOption name="ca_list_file">
767 <synopsis>File containing a list of certificates to read (TLS ONLY)</synopsis>
769 <configOption name="cert_file">
770 <synopsis>Certificate file for endpoint (TLS ONLY)</synopsis>
772 <configOption name="cipher">
773 <synopsis>Preferred Cryptography Cipher (TLS ONLY)</synopsis>
775 Many options for acceptable ciphers see link for more:
776 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
777 </para></description>
779 <configOption name="domain">
780 <synopsis>Domain the transport comes from</synopsis>
782 <configOption name="external_media_address">
783 <synopsis>External IP address to use in RTP handling</synopsis>
785 When a request or response is sent out, if the destination of the
786 message is outside the IP network defined in the option <literal>localnet</literal>,
787 and the media address in the SDP is within the localnet network, then the
788 media address in the SDP will be rewritten to the value defined for
789 <literal>external_media_address</literal>.
790 </para></description>
792 <configOption name="external_signaling_address">
793 <synopsis>External address for SIP signalling</synopsis>
795 <configOption name="external_signaling_port" default="0">
796 <synopsis>External port for SIP signalling</synopsis>
798 <configOption name="method">
799 <synopsis>Method of SSL transport (TLS ONLY)</synopsis>
802 <enum name="default" />
803 <enum name="unspecified" />
804 <enum name="tlsv1" />
805 <enum name="sslv2" />
806 <enum name="sslv3" />
807 <enum name="sslv23" />
811 <configOption name="local_net">
812 <synopsis>Network to consider local (used for NAT purposes).</synopsis>
813 <description><para>This must be in CIDR or dotted decimal format with the IP
814 and mask separated with a slash ('/').</para></description>
816 <configOption name="password">
817 <synopsis>Password required for transport</synopsis>
819 <configOption name="priv_key_file">
820 <synopsis>Private key file (TLS ONLY)</synopsis>
822 <configOption name="protocol" default="udp">
823 <synopsis>Protocol to use for SIP traffic</synopsis>
834 <configOption name="require_client_cert" default="false">
835 <synopsis>Require client certificate (TLS ONLY)</synopsis>
837 <configOption name="type">
838 <synopsis>Must be of type 'transport'.</synopsis>
840 <configOption name="verify_client" default="false">
841 <synopsis>Require verification of client certificate (TLS ONLY)</synopsis>
843 <configOption name="verify_server" default="false">
844 <synopsis>Require verification of server certificate (TLS ONLY)</synopsis>
846 <configOption name="tos" default="false">
847 <synopsis>Enable TOS for the signalling sent over this transport</synopsis>
849 <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
850 for more information on this parameter.</para>
851 <note><para>This option does not apply to the <replaceable>ws</replaceable>
852 or the <replaceable>wss</replaceable> protocols.</para></note>
855 <configOption name="cos" default="false">
856 <synopsis>Enable COS for the signalling sent over this transport</synopsis>
858 <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
859 for more information on this parameter.</para>
860 <note><para>This option does not apply to the <replaceable>ws</replaceable>
861 or the <replaceable>wss</replaceable> protocols.</para></note>
865 <configObject name="contact">
866 <synopsis>A way of creating an aliased name to a SIP URI</synopsis>
868 Contacts are a way to hide SIP URIs from the dialplan directly.
869 They are also used to make a group of contactable parties when
870 in use with <literal>AoR</literal> lists.
871 </para></description>
872 <configOption name="type">
873 <synopsis>Must be of type 'contact'.</synopsis>
875 <configOption name="uri">
876 <synopsis>SIP URI to contact peer</synopsis>
878 <configOption name="expiration_time">
879 <synopsis>Time to keep alive a contact</synopsis>
881 Time to keep alive a contact. String style specification.
882 </para></description>
884 <configOption name="qualify_frequency" default="0">
885 <synopsis>Interval at which to qualify a contact</synopsis>
887 Interval between attempts to qualify the contact for reachability.
888 If <literal>0</literal> never qualify. Time in seconds.
889 </para></description>
891 <configOption name="outbound_proxy">
892 <synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
894 If set the provided URI will be used as the outbound proxy when an
895 OPTIONS request is sent to a contact for qualify purposes.
896 </para></description>
898 <configOption name="path">
899 <synopsis>Stored Path vector for use in Route headers on outgoing requests.</synopsis>
901 <configOption name="user_agent">
902 <synopsis>User-Agent header from registration.</synopsis>
904 The User-Agent is automatically stored based on data present in incoming SIP
905 REGISTER requests and is not intended to be configured manually.
906 </para></description>
909 <configObject name="aor">
910 <synopsis>The configuration for a location of an endpoint</synopsis>
912 An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
913 AoRs are specified, an endpoint will not be reachable by Asterisk.
914 Beyond that, an AoR has other uses within Asterisk, such as inbound
917 An <literal>AoR</literal> is a way to allow dialing a group
918 of <literal>Contacts</literal> that all use the same
919 <literal>endpoint</literal> for calls.
921 This can be used as another way of grouping a list of contacts to dial
922 rather than specifing them each directly when dialing via the dialplan.
923 This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
925 Registrations: For Asterisk to match an inbound registration to an endpoint,
926 the AoR object name must match the user portion of the SIP URI in the "To:"
927 header of the inbound SIP registration. That will usually be equivalent
928 to the "user name" set in your hard or soft phones configuration.
929 </para></description>
930 <configOption name="contact">
931 <synopsis>Permanent contacts assigned to AoR</synopsis>
933 Contacts specified will be called whenever referenced
934 by <literal>chan_pjsip</literal>.
936 Use a separate "contact=" entry for each contact required. Contacts
937 are specified using a SIP URI.
938 </para></description>
940 <configOption name="default_expiration" default="3600">
941 <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
943 <configOption name="mailboxes">
944 <synopsis>Allow subscriptions for the specified mailbox(es)</synopsis>
945 <description><para>This option applies when an external entity subscribes to an AoR
946 for Message Waiting Indications. The mailboxes specified will be subscribed to.
947 More than one mailbox can be specified with a comma-delimited string.
948 app_voicemail mailboxes must be specified as mailbox@context;
949 for example: mailboxes=6001@default. For mailboxes provided by external sources,
950 such as through the res_external_mwi module, you must specify strings supported by
953 For endpoints that cannot SUBSCRIBE for MWI, you can set the <literal>mailboxes</literal> option in your
954 endpoint configuration section to enable unsolicited MWI NOTIFYs to the endpoint.
955 </para></description>
957 <configOption name="maximum_expiration" default="7200">
958 <synopsis>Maximum time to keep an AoR</synopsis>
960 Maximium time to keep a peer with explicit expiration. Time in seconds.
961 </para></description>
963 <configOption name="max_contacts" default="0">
964 <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
966 Maximum number of contacts that can associate with this AoR. This value does
967 not affect the number of contacts that can be added with the "contact" option.
968 It only limits contacts added through external interaction, such as
971 <note><para>This should be set to <literal>1</literal> and
972 <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
973 wish to stick with the older <literal>chan_sip</literal> behaviour.
977 <configOption name="minimum_expiration" default="60">
978 <synopsis>Minimum keep alive time for an AoR</synopsis>
980 Minimum time to keep a peer with an explict expiration. Time in seconds.
981 </para></description>
983 <configOption name="remove_existing" default="no">
984 <synopsis>Determines whether new contacts replace existing ones.</synopsis>
986 On receiving a new registration to the AoR should it remove
987 the existing contact that was registered against it?
989 <note><para>This should be set to <literal>yes</literal> and
990 <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
991 wish to stick with the older <literal>chan_sip</literal> behaviour.
995 <configOption name="type">
996 <synopsis>Must be of type 'aor'.</synopsis>
998 <configOption name="qualify_frequency" default="0">
999 <synopsis>Interval at which to qualify an AoR</synopsis>
1001 Interval between attempts to qualify the AoR for reachability.
1002 If <literal>0</literal> never qualify. Time in seconds.
1003 </para></description>
1005 <configOption name="authenticate_qualify" default="no">
1006 <synopsis>Authenticates a qualify request if needed</synopsis>
1008 If true and a qualify request receives a challenge or authenticate response
1009 authentication is attempted before declaring the contact available.
1010 </para></description>
1012 <configOption name="outbound_proxy">
1013 <synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
1015 If set the provided URI will be used as the outbound proxy when an
1016 OPTIONS request is sent to a contact for qualify purposes.
1017 </para></description>
1019 <configOption name="support_path">
1020 <synopsis>Enables Path support for REGISTER requests and Route support for other requests.</synopsis>
1022 When this option is enabled, the Path headers in register requests will be saved
1023 and its contents will be used in Route headers for outbound out-of-dialog requests
1024 and in Path headers for outbound 200 responses. Path support will also be indicated
1025 in the Supported header.
1026 </para></description>
1029 <configObject name="system">
1030 <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
1032 The settings in this section are global. In addition to being global, the values will
1033 not be re-evaluated when a reload is performed. This is because the values must be set
1034 before the SIP stack is initialized. The only way to reset these values is to either
1035 restart Asterisk, or unload res_pjsip.so and then load it again.
1036 </para></description>
1037 <configOption name="timer_t1" default="500">
1038 <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
1040 Timer T1 is the base for determining how long to wait before retransmitting
1041 requests that receive no response when using an unreliable transport (e.g. UDP).
1042 For more information on this timer, see RFC 3261, Section 17.1.1.1.
1043 </para></description>
1045 <configOption name="timer_b" default="32000">
1046 <synopsis>Set transaction timer B value (milliseconds).</synopsis>
1048 Timer B determines the maximum amount of time to wait after sending an INVITE
1049 request before terminating the transaction. It is recommended that this be set
1050 to 64 * Timer T1, but it may be set higher if desired. For more information on
1051 this timer, see RFC 3261, Section 17.1.1.1.
1052 </para></description>
1054 <configOption name="compact_headers" default="no">
1055 <synopsis>Use the short forms of common SIP header names.</synopsis>
1057 <configOption name="threadpool_initial_size" default="0">
1058 <synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
1060 <configOption name="threadpool_auto_increment" default="5">
1061 <synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
1063 <configOption name="threadpool_idle_timeout" default="60">
1064 <synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
1066 <configOption name="threadpool_max_size" default="0">
1067 <synopsis>Maximum number of threads in the res_pjsip threadpool.
1068 A value of 0 indicates no maximum.</synopsis>
1070 <configOption name="type">
1071 <synopsis>Must be of type 'system'.</synopsis>
1074 <configObject name="global">
1075 <synopsis>Options that apply globally to all SIP communications</synopsis>
1077 The settings in this section are global. Unlike options in the <literal>system</literal>
1078 section, these options can be refreshed by performing a reload.
1079 </para></description>
1080 <configOption name="max_forwards" default="70">
1081 <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
1083 <configOption name="type">
1084 <synopsis>Must be of type 'global'.</synopsis>
1086 <configOption name="user_agent" default="Asterisk <Asterisk Version>">
1087 <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
1089 <configOption name="default_outbound_endpoint" default="default_outbound_endpoint">
1090 <synopsis>Endpoint to use when sending an outbound request to a URI without a specified endpoint.</synopsis>
1092 <configOption name="debug" default="no">
1093 <synopsis>Enable/Disable SIP debug logging. Valid options include yes|no or
1094 a host address</synopsis>
1099 <manager name="PJSIPQualify" language="en_US">
1101 Qualify a chan_pjsip endpoint.
1104 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1105 <parameter name="Endpoint" required="true">
1106 <para>The endpoint you want to qualify.</para>
1110 <para>Qualify a chan_pjsip endpoint.</para>
1113 <manager name="PJSIPShowEndpoints" language="en_US">
1115 Lists PJSIP endpoints.
1120 Provides a listing of all endpoints. For each endpoint an <literal>EndpointList</literal> event
1121 is raised that contains relevant attributes and status information. Once all
1122 endpoints have been listed an <literal>EndpointListComplete</literal> event is issued.
1126 <manager name="PJSIPShowEndpoint" language="en_US">
1128 Detail listing of an endpoint and its objects.
1131 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1132 <parameter name="Endpoint" required="true">
1133 <para>The endpoint to list.</para>
1138 Provides a detailed listing of options for a given endpoint. Events are issued
1139 showing the configuration and status of the endpoint and associated objects. These
1140 events include <literal>EndpointDetail</literal>, <literal>AorDetail</literal>,
1141 <literal>AuthDetail</literal>, <literal>TransportDetail</literal>, and
1142 <literal>IdentifyDetail</literal>. Some events may be listed multiple times if multiple objects are
1143 associated (for instance AoRs). Once all detail events have been raised a final
1144 <literal>EndpointDetailComplete</literal> event is issued.
1150 #define MOD_DATA_CONTACT "contact"
1152 static pjsip_endpoint *ast_pjsip_endpoint;
1154 static struct ast_threadpool *sip_threadpool;
1156 static int register_service(void *data)
1158 pjsip_module **module = data;
1159 if (!ast_pjsip_endpoint) {
1160 ast_log(LOG_ERROR, "There is no PJSIP endpoint. Unable to register services\n");
1163 if (pjsip_endpt_register_module(ast_pjsip_endpoint, *module) != PJ_SUCCESS) {
1164 ast_log(LOG_ERROR, "Unable to register module %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1167 ast_debug(1, "Registered SIP service %.*s (%p)\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name), *module);
1168 ast_module_ref(ast_module_info->self);
1172 int ast_sip_register_service(pjsip_module *module)
1174 return ast_sip_push_task_synchronous(NULL, register_service, &module);
1177 static int unregister_service(void *data)
1179 pjsip_module **module = data;
1180 ast_module_unref(ast_module_info->self);
1181 if (!ast_pjsip_endpoint) {
1184 pjsip_endpt_unregister_module(ast_pjsip_endpoint, *module);
1185 ast_debug(1, "Unregistered SIP service %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1189 void ast_sip_unregister_service(pjsip_module *module)
1191 ast_sip_push_task_synchronous(NULL, unregister_service, &module);
1194 static struct ast_sip_authenticator *registered_authenticator;
1196 int ast_sip_register_authenticator(struct ast_sip_authenticator *auth)
1198 if (registered_authenticator) {
1199 ast_log(LOG_WARNING, "Authenticator %p is already registered. Cannot register a new one\n", registered_authenticator);
1202 registered_authenticator = auth;
1203 ast_debug(1, "Registered SIP authenticator module %p\n", auth);
1204 ast_module_ref(ast_module_info->self);
1208 void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth)
1210 if (registered_authenticator != auth) {
1211 ast_log(LOG_WARNING, "Trying to unregister authenticator %p but authenticator %p registered\n",
1212 auth, registered_authenticator);
1215 registered_authenticator = NULL;
1216 ast_debug(1, "Unregistered SIP authenticator %p\n", auth);
1217 ast_module_unref(ast_module_info->self);
1220 int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1222 if (!registered_authenticator) {
1223 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is not required\n");
1227 return registered_authenticator->requires_authentication(endpoint, rdata);
1230 enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
1231 pjsip_rx_data *rdata, pjsip_tx_data *tdata)
1233 if (!registered_authenticator) {
1234 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is successful\n");
1237 return registered_authenticator->check_authentication(endpoint, rdata, tdata);
1240 static struct ast_sip_outbound_authenticator *registered_outbound_authenticator;
1242 int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1244 if (registered_outbound_authenticator) {
1245 ast_log(LOG_WARNING, "Outbound authenticator %p is already registered. Cannot register a new one\n", registered_outbound_authenticator);
1248 registered_outbound_authenticator = auth;
1249 ast_debug(1, "Registered SIP outbound authenticator module %p\n", auth);
1250 ast_module_ref(ast_module_info->self);
1254 void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1256 if (registered_outbound_authenticator != auth) {
1257 ast_log(LOG_WARNING, "Trying to unregister outbound authenticator %p but outbound authenticator %p registered\n",
1258 auth, registered_outbound_authenticator);
1261 registered_outbound_authenticator = NULL;
1262 ast_debug(1, "Unregistered SIP outbound authenticator %p\n", auth);
1263 ast_module_unref(ast_module_info->self);
1266 int ast_sip_create_request_with_auth(const struct ast_sip_auth_vector *auths, pjsip_rx_data *challenge,
1267 pjsip_transaction *tsx, pjsip_tx_data **new_request)
1269 if (!registered_outbound_authenticator) {
1270 ast_log(LOG_WARNING, "No SIP outbound authenticator registered. Cannot respond to authentication challenge\n");
1273 return registered_outbound_authenticator->create_request_with_auth(auths, challenge, tsx, new_request);
1276 struct endpoint_identifier_list {
1277 struct ast_sip_endpoint_identifier *identifier;
1278 AST_RWLIST_ENTRY(endpoint_identifier_list) list;
1281 static AST_RWLIST_HEAD_STATIC(endpoint_identifiers, endpoint_identifier_list);
1283 int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1285 struct endpoint_identifier_list *id_list_item;
1286 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1288 id_list_item = ast_calloc(1, sizeof(*id_list_item));
1289 if (!id_list_item) {
1290 ast_log(LOG_ERROR, "Unabled to add endpoint identifier. Out of memory.\n");
1293 id_list_item->identifier = identifier;
1295 AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list);
1296 ast_debug(1, "Registered endpoint identifier %p\n", identifier);
1298 ast_module_ref(ast_module_info->self);
1302 void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1304 struct endpoint_identifier_list *iter;
1305 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1306 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) {
1307 if (iter->identifier == identifier) {
1308 AST_RWLIST_REMOVE_CURRENT(list);
1310 ast_debug(1, "Unregistered endpoint identifier %p\n", identifier);
1311 ast_module_unref(ast_module_info->self);
1315 AST_RWLIST_TRAVERSE_SAFE_END;
1318 struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata)
1320 struct endpoint_identifier_list *iter;
1321 struct ast_sip_endpoint *endpoint = NULL;
1322 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1323 AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) {
1324 ast_assert(iter->identifier->identify_endpoint != NULL);
1325 endpoint = iter->identifier->identify_endpoint(rdata);
1333 AST_RWLIST_HEAD_STATIC(endpoint_formatters, ast_sip_endpoint_formatter);
1335 int ast_sip_register_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
1337 SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1338 AST_RWLIST_INSERT_TAIL(&endpoint_formatters, obj, next);
1339 ast_module_ref(ast_module_info->self);
1343 void ast_sip_unregister_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
1345 struct ast_sip_endpoint_formatter *i;
1346 SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1347 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_formatters, i, next) {
1349 AST_RWLIST_REMOVE_CURRENT(next);
1350 ast_module_unref(ast_module_info->self);
1354 AST_RWLIST_TRAVERSE_SAFE_END;
1357 int ast_sip_format_endpoint_ami(struct ast_sip_endpoint *endpoint,
1358 struct ast_sip_ami *ami, int *count)
1361 struct ast_sip_endpoint_formatter *i;
1362 SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1364 AST_RWLIST_TRAVERSE(&endpoint_formatters, i, next) {
1365 if (i->format_ami && ((res = i->format_ami(endpoint, ami)) < 0)) {
1376 pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
1378 return ast_pjsip_endpoint;
1381 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)
1383 pj_str_t tmp, local_addr;
1385 pjsip_sip_uri *sip_uri;
1386 pjsip_transport_type_e type = PJSIP_TRANSPORT_UNSPECIFIED;
1388 char uuid_str[AST_UUID_STR_LEN];
1390 if (ast_strlen_zero(user)) {
1391 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1395 user = ast_uuid_to_str(uuid, uuid_str, sizeof(uuid_str));
1398 /* Parse the provided target URI so we can determine what transport it will end up using */
1399 pj_strdup_with_null(pool, &tmp, target);
1401 if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) ||
1402 (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
1406 sip_uri = pjsip_uri_get_uri(uri);
1408 /* Determine the transport type to use */
1409 if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
1410 type = PJSIP_TRANSPORT_TLS;
1411 } else if (!sip_uri->transport_param.slen) {
1412 type = PJSIP_TRANSPORT_UDP;
1414 type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
1417 if (type == PJSIP_TRANSPORT_UNSPECIFIED) {
1421 /* If the host is IPv6 turn the transport into an IPv6 version */
1422 if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1423 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1426 if (!ast_strlen_zero(domain)) {
1427 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1428 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1432 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1433 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1437 /* Get the local bound address for the transport that will be used when communicating with the provided URI */
1438 if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), pool, type, selector,
1439 &local_addr, &local_port) != PJ_SUCCESS) {
1441 /* If no local address can be retrieved using the transport manager use the host one */
1442 pj_strdup(pool, &local_addr, pj_gethostname());
1443 local_port = pjsip_transport_get_default_port_for_type(PJSIP_TRANSPORT_UDP);
1446 /* If IPv6 was specified in the transport, set the proper type */
1447 if (pj_strchr(&local_addr, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1448 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1451 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1452 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1453 "<sip:%s@%s%.*s%s:%d%s%s>",
1455 (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1456 (int)local_addr.slen,
1458 (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1460 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1461 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1466 static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
1468 RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
1469 const char *transport_name = endpoint->transport;
1471 if (ast_strlen_zero(transport_name)) {
1475 transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name);
1477 if (!transport || !transport->state) {
1478 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport '%s' for endpoint '%s'\n",
1479 transport_name, ast_sorcery_object_get_id(endpoint));
1483 if (transport->state->transport) {
1484 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1485 selector->u.transport = transport->state->transport;
1486 } else if (transport->state->factory) {
1487 selector->type = PJSIP_TPSELECTOR_LISTENER;
1488 selector->u.listener = transport->state->factory;
1496 pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
1498 char enclosed_uri[PJSIP_MAX_URL_SIZE];
1499 pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri, target_uri;
1500 pjsip_dialog *dlg = NULL;
1501 const char *outbound_proxy = endpoint->outbound_proxy;
1502 pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1503 static const pj_str_t HCONTACT = { "Contact", 7 };
1505 snprintf(enclosed_uri, sizeof(enclosed_uri), "<%s>", uri);
1506 pj_cstr(&remote_uri, enclosed_uri);
1508 pj_cstr(&target_uri, uri);
1510 if (pjsip_dlg_create_uac(pjsip_ua_instance(), &local_uri, NULL, &remote_uri, &target_uri, &dlg) != PJ_SUCCESS) {
1514 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1515 pjsip_dlg_terminate(dlg);
1519 if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
1520 pjsip_dlg_terminate(dlg);
1524 /* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
1525 pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
1526 dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
1527 dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
1529 /* If a request user has been specified and we are permitted to change it, do so */
1530 if (!ast_strlen_zero(request_user)) {
1531 pjsip_sip_uri *sip_uri;
1533 if (PJSIP_URI_SCHEME_IS_SIP(dlg->target) || PJSIP_URI_SCHEME_IS_SIPS(dlg->target)) {
1534 sip_uri = pjsip_uri_get_uri(dlg->target);
1535 pj_strdup2(dlg->pool, &sip_uri->user, request_user);
1537 if (PJSIP_URI_SCHEME_IS_SIP(dlg->remote.info->uri) || PJSIP_URI_SCHEME_IS_SIPS(dlg->remote.info->uri)) {
1538 sip_uri = pjsip_uri_get_uri(dlg->remote.info->uri);
1539 pj_strdup2(dlg->pool, &sip_uri->user, request_user);
1543 /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
1546 pjsip_dlg_set_transport(dlg, &selector);
1548 if (!ast_strlen_zero(outbound_proxy)) {
1549 pjsip_route_hdr route_set, *route;
1550 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1553 pj_list_init(&route_set);
1555 pj_strdup2_with_null(dlg->pool, &tmp, outbound_proxy);
1556 if (!(route = pjsip_parse_hdr(dlg->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1558 pjsip_dlg_terminate(dlg);
1561 pj_list_insert_nodes_before(&route_set, route);
1563 pjsip_dlg_set_route_set(dlg, &route_set);
1571 pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1575 pjsip_transport_type_e type = rdata->tp_info.transport->key.type;
1578 contact.ptr = pj_pool_alloc(rdata->tp_info.pool, PJSIP_MAX_URL_SIZE);
1579 contact.slen = pj_ansi_snprintf(contact.ptr, PJSIP_MAX_URL_SIZE,
1580 "<sip:%s%.*s%s:%d%s%s>",
1581 (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1582 (int)rdata->tp_info.transport->local_name.host.slen,
1583 rdata->tp_info.transport->local_name.host.ptr,
1584 (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1585 rdata->tp_info.transport->local_name.port,
1586 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1587 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1589 status = pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, &contact, &dlg);
1590 if (status != PJ_SUCCESS) {
1591 char err[PJ_ERR_MSG_SIZE];
1593 pj_strerror(status, err, sizeof(err));
1594 ast_log(LOG_ERROR, "Could not create dialog with endpoint %s. %s\n",
1595 ast_sorcery_object_get_id(endpoint), err);
1602 /* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
1603 static const pjsip_method info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
1604 static const pjsip_method message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
1608 const pjsip_method *pmethod;
1610 { "INVITE", &pjsip_invite_method },
1611 { "CANCEL", &pjsip_cancel_method },
1612 { "ACK", &pjsip_ack_method },
1613 { "BYE", &pjsip_bye_method },
1614 { "REGISTER", &pjsip_register_method },
1615 { "OPTIONS", &pjsip_options_method },
1616 { "SUBSCRIBE", &pjsip_subscribe_method },
1617 { "NOTIFY", &pjsip_notify_method },
1618 { "PUBLISH", &pjsip_publish_method },
1619 { "INFO", &info_method },
1620 { "MESSAGE", &message_method },
1623 static const pjsip_method *get_pjsip_method(const char *method)
1626 for (i = 0; i < ARRAY_LEN(methods); ++i) {
1627 if (!strcmp(method, methods[i].method)) {
1628 return methods[i].pmethod;
1634 static int create_in_dialog_request(const pjsip_method *method, struct pjsip_dialog *dlg, pjsip_tx_data **tdata)
1636 if (pjsip_dlg_create_request(dlg, method, -1, tdata) != PJ_SUCCESS) {
1637 ast_log(LOG_WARNING, "Unable to create in-dialog request.\n");
1644 static pj_bool_t supplement_on_rx_request(pjsip_rx_data *rdata);
1645 static pjsip_module supplement_module = {
1646 .name = { "Out of dialog supplement hook", 29 },
1648 .priority = PJSIP_MOD_PRIORITY_APPLICATION - 1,
1649 .on_rx_request = supplement_on_rx_request,
1652 static int create_out_of_dialog_request(const pjsip_method *method, struct ast_sip_endpoint *endpoint,
1653 const char *uri, struct ast_sip_contact *provided_contact, pjsip_tx_data **tdata)
1655 RAII_VAR(struct ast_sip_contact *, contact, ao2_bump(provided_contact), ao2_cleanup);
1656 pj_str_t remote_uri;
1659 pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1661 if (ast_strlen_zero(uri)) {
1662 if (!endpoint && !contact) {
1663 ast_log(LOG_ERROR, "An endpoint and/or uri must be specified\n");
1668 contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
1670 if (!contact || ast_strlen_zero(contact->uri)) {
1671 ast_log(LOG_ERROR, "Unable to retrieve contact for endpoint %s\n",
1672 ast_sorcery_object_get_id(endpoint));
1676 pj_cstr(&remote_uri, contact->uri);
1678 pj_cstr(&remote_uri, uri);
1682 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1683 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
1684 ast_sorcery_object_get_id(endpoint));
1689 pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
1692 ast_log(LOG_ERROR, "Unable to create PJLIB memory pool\n");
1696 if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL,
1697 endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
1698 ast_log(LOG_ERROR, "Unable to create From header for %.*s request to endpoint %s\n",
1699 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1700 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1704 if (pjsip_endpt_create_request(ast_sip_get_pjsip_endpoint(), method, &remote_uri,
1705 &from, &remote_uri, &from, NULL, -1, NULL, tdata) != PJ_SUCCESS) {
1706 ast_log(LOG_ERROR, "Unable to create outbound %.*s request to endpoint %s\n",
1707 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1708 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1712 /* If an outbound proxy is specified on the endpoint apply it to this request */
1713 if (endpoint && !ast_strlen_zero(endpoint->outbound_proxy) &&
1714 ast_sip_set_outbound_proxy((*tdata), endpoint->outbound_proxy)) {
1715 ast_log(LOG_ERROR, "Unable to apply outbound proxy on request %.*s to endpoint %s\n",
1716 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1717 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1721 ast_sip_mod_data_set((*tdata)->pool, (*tdata)->mod_data, supplement_module.id, MOD_DATA_CONTACT, ao2_bump(contact));
1723 /* We can release this pool since request creation copied all the necessary
1724 * data into the outbound request's pool
1726 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1730 int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
1731 struct ast_sip_endpoint *endpoint, const char *uri,
1732 struct ast_sip_contact *contact, pjsip_tx_data **tdata)
1734 const pjsip_method *pmethod = get_pjsip_method(method);
1737 ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
1742 return create_in_dialog_request(pmethod, dlg, tdata);
1744 return create_out_of_dialog_request(pmethod, endpoint, uri, contact, tdata);
1748 AST_RWLIST_HEAD_STATIC(supplements, ast_sip_supplement);
1750 int ast_sip_register_supplement(struct ast_sip_supplement *supplement)
1752 struct ast_sip_supplement *iter;
1754 SCOPED_LOCK(lock, &supplements, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1756 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&supplements, iter, next) {
1757 if (iter->priority > supplement->priority) {
1758 AST_RWLIST_INSERT_BEFORE_CURRENT(supplement, next);
1763 AST_RWLIST_TRAVERSE_SAFE_END;
1766 AST_RWLIST_INSERT_TAIL(&supplements, supplement, next);
1768 ast_module_ref(ast_module_info->self);
1772 void ast_sip_unregister_supplement(struct ast_sip_supplement *supplement)
1774 struct ast_sip_supplement *iter;
1775 SCOPED_LOCK(lock, &supplements, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1776 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&supplements, iter, next) {
1777 if (supplement == iter) {
1778 AST_RWLIST_REMOVE_CURRENT(next);
1779 ast_module_unref(ast_module_info->self);
1783 AST_RWLIST_TRAVERSE_SAFE_END;
1786 static int send_in_dialog_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg)
1788 if (pjsip_dlg_send_request(dlg, tdata, -1, NULL) != PJ_SUCCESS) {
1789 ast_log(LOG_WARNING, "Unable to send in-dialog request.\n");
1795 static pj_bool_t does_method_match(const pj_str_t *message_method, const char *supplement_method)
1799 if (ast_strlen_zero(supplement_method)) {
1803 pj_cstr(&method, supplement_method);
1805 return pj_stristr(&method, message_method) ? PJ_TRUE : PJ_FALSE;
1808 /*! \brief Structure to hold information about an outbound request */
1809 struct send_request_data {
1810 struct ast_sip_endpoint *endpoint; /*! The endpoint associated with this request */
1811 void *token; /*! Information to be provided to the callback upon receipt of a response */
1812 void (*callback)(void *token, pjsip_event *e); /*! The callback to be called upon receipt of a response */
1815 static void send_request_data_destroy(void *obj)
1817 struct send_request_data *req_data = obj;
1818 ao2_cleanup(req_data->endpoint);
1821 static struct send_request_data *send_request_data_alloc(struct ast_sip_endpoint *endpoint,
1822 void *token, void (*callback)(void *token, pjsip_event *e))
1824 struct send_request_data *req_data = ao2_alloc(sizeof(*req_data), send_request_data_destroy);
1830 req_data->endpoint = ao2_bump(endpoint);
1831 req_data->token = token;
1832 req_data->callback = callback;
1837 static void send_request_cb(void *token, pjsip_event *e)
1839 RAII_VAR(struct send_request_data *, req_data, token, ao2_cleanup);
1840 pjsip_transaction *tsx = e->body.tsx_state.tsx;
1841 pjsip_rx_data *challenge = e->body.tsx_state.src.rdata;
1842 pjsip_tx_data *tdata;
1843 struct ast_sip_supplement *supplement;
1845 AST_RWLIST_RDLOCK(&supplements);
1846 AST_LIST_TRAVERSE(&supplements, supplement, next) {
1847 if (supplement->incoming_response && does_method_match(&challenge->msg_info.cseq->method.name, supplement->method)) {
1848 supplement->incoming_response(req_data->endpoint, challenge);
1851 AST_RWLIST_UNLOCK(&supplements);
1853 if (tsx->status_code == 401 || tsx->status_code == 407) {
1854 if (!ast_sip_create_request_with_auth(&req_data->endpoint->outbound_auths, challenge, tsx, &tdata)) {
1855 pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, req_data->token, req_data->callback);
1860 if (req_data->callback) {
1861 req_data->callback(req_data->token, e);
1865 static int send_out_of_dialog_request(pjsip_tx_data *tdata, struct ast_sip_endpoint *endpoint,
1866 void *token, void (*callback)(void *token, pjsip_event *e))
1868 struct ast_sip_supplement *supplement;
1869 struct send_request_data *req_data = send_request_data_alloc(endpoint, token, callback);
1870 struct ast_sip_contact *contact = ast_sip_mod_data_get(tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT);
1876 AST_RWLIST_RDLOCK(&supplements);
1877 AST_LIST_TRAVERSE(&supplements, supplement, next) {
1878 if (supplement->outgoing_request && does_method_match(&tdata->msg->line.req.method.name, supplement->method)) {
1879 supplement->outgoing_request(endpoint, contact, tdata);
1882 AST_RWLIST_UNLOCK(&supplements);
1884 ast_sip_mod_data_set(tdata->pool, tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT, NULL);
1885 ao2_cleanup(contact);
1887 if (pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, req_data, send_request_cb) != PJ_SUCCESS) {
1888 ast_log(LOG_ERROR, "Error attempting to send outbound %.*s request to endpoint %s\n",
1889 (int) pj_strlen(&tdata->msg->line.req.method.name),
1890 pj_strbuf(&tdata->msg->line.req.method.name),
1891 ast_sorcery_object_get_id(endpoint));
1892 ao2_cleanup(req_data);
1899 int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg,
1900 struct ast_sip_endpoint *endpoint, void *token,
1901 void (*callback)(void *token, pjsip_event *e))
1903 ast_assert(tdata->msg->type == PJSIP_REQUEST_MSG);
1906 return send_in_dialog_request(tdata, dlg);
1908 return send_out_of_dialog_request(tdata, endpoint, token, callback);
1912 int ast_sip_set_outbound_proxy(pjsip_tx_data *tdata, const char *proxy)
1914 pjsip_route_hdr *route;
1915 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1918 pj_strdup2_with_null(tdata->pool, &tmp, proxy);
1919 if (!(route = pjsip_parse_hdr(tdata->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1923 pj_list_insert_nodes_before(&tdata->msg->hdr, (pjsip_hdr*)route);
1928 int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value)
1932 pjsip_generic_string_hdr *hdr;
1934 pj_cstr(&hdr_name, name);
1935 pj_cstr(&hdr_value, value);
1937 hdr = pjsip_generic_string_hdr_create(tdata->pool, &hdr_name, &hdr_value);
1939 pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) hdr);
1943 static pjsip_msg_body *ast_body_to_pjsip_body(pj_pool_t *pool, const struct ast_sip_body *body)
1949 pj_cstr(&type, body->type);
1950 pj_cstr(&subtype, body->subtype);
1951 pj_cstr(&body_text, body->body_text);
1953 return pjsip_msg_body_create(pool, &type, &subtype, &body_text);
1956 int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body)
1958 pjsip_msg_body *pjsip_body = ast_body_to_pjsip_body(tdata->pool, body);
1959 tdata->msg->body = pjsip_body;
1963 int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies)
1966 /* NULL for type and subtype automatically creates "multipart/mixed" */
1967 pjsip_msg_body *body = pjsip_multipart_create(tdata->pool, NULL, NULL);
1969 for (i = 0; i < num_bodies; ++i) {
1970 pjsip_multipart_part *part = pjsip_multipart_create_part(tdata->pool);
1971 part->body = ast_body_to_pjsip_body(tdata->pool, bodies[i]);
1972 pjsip_multipart_add_part(tdata->pool, body, part);
1975 tdata->msg->body = body;
1979 int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text)
1981 size_t combined_size = strlen(body_text) + tdata->msg->body->len;
1982 struct ast_str *body_buffer = ast_str_alloca(combined_size);
1984 ast_str_set(&body_buffer, 0, "%.*s%s", (int) tdata->msg->body->len, (char *) tdata->msg->body->data, body_text);
1986 tdata->msg->body->data = pj_pool_alloc(tdata->pool, combined_size);
1987 pj_memcpy(tdata->msg->body->data, ast_str_buffer(body_buffer), combined_size);
1988 tdata->msg->body->len = combined_size;
1993 struct ast_taskprocessor *ast_sip_create_serializer(void)
1995 struct ast_taskprocessor *serializer;
1996 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1997 char name[AST_UUID_STR_LEN];
2003 ast_uuid_to_str(uuid, name, sizeof(name));
2005 serializer = ast_threadpool_serializer(name, sip_threadpool);
2012 int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
2015 return ast_taskprocessor_push(serializer, sip_task, task_data);
2017 return ast_threadpool_push(sip_threadpool, sip_task, task_data);
2021 struct sync_task_data {
2026 int (*task)(void *);
2030 static int sync_task(void *data)
2032 struct sync_task_data *std = data;
2033 std->fail = std->task(std->task_data);
2035 ast_mutex_lock(&std->lock);
2037 ast_cond_signal(&std->cond);
2038 ast_mutex_unlock(&std->lock);
2042 int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
2044 /* This method is an onion */
2045 struct sync_task_data std;
2047 if (ast_sip_thread_is_servant()) {
2048 return sip_task(task_data);
2051 ast_mutex_init(&std.lock);
2052 ast_cond_init(&std.cond, NULL);
2053 std.fail = std.complete = 0;
2054 std.task = sip_task;
2055 std.task_data = task_data;
2058 if (ast_taskprocessor_push(serializer, sync_task, &std)) {
2062 if (ast_threadpool_push(sip_threadpool, sync_task, &std)) {
2067 ast_mutex_lock(&std.lock);
2068 while (!std.complete) {
2069 ast_cond_wait(&std.cond, &std.lock);
2071 ast_mutex_unlock(&std.lock);
2073 ast_mutex_destroy(&std.lock);
2074 ast_cond_destroy(&std.cond);
2078 void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
2080 size_t chars_to_copy = MIN(size - 1, pj_strlen(src));
2081 memcpy(dest, pj_strbuf(src), chars_to_copy);
2082 dest[chars_to_copy] = '\0';
2085 int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype)
2087 pjsip_media_type compare;
2089 if (!content_type) {
2093 pjsip_media_type_init2(&compare, type, subtype);
2095 return pjsip_media_type_cmp(content_type, &compare, 0) ? 0 : -1;
2098 pj_caching_pool caching_pool;
2099 pj_pool_t *memory_pool;
2100 pj_thread_t *monitor_thread;
2101 static int monitor_continue;
2103 static void *monitor_thread_exec(void *endpt)
2105 while (monitor_continue) {
2106 const pj_time_val delay = {0, 10};
2107 pjsip_endpt_handle_events(ast_pjsip_endpoint, &delay);
2112 static void stop_monitor_thread(void)
2114 monitor_continue = 0;
2115 pj_thread_join(monitor_thread);
2118 AST_THREADSTORAGE(pj_thread_storage);
2119 AST_THREADSTORAGE(servant_id_storage);
2120 #define SIP_SERVANT_ID 0x5E2F1D
2122 static void sip_thread_start(void)
2124 pj_thread_desc *desc;
2125 pj_thread_t *thread;
2126 uint32_t *servant_id;
2128 servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
2130 ast_log(LOG_ERROR, "Could not set SIP servant ID in thread-local storage.\n");
2133 *servant_id = SIP_SERVANT_ID;
2135 desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc));
2137 ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n");
2140 pj_bzero(*desc, sizeof(*desc));
2142 if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) {
2143 ast_log(LOG_ERROR, "Couldn't register thread with PJLIB.\n");
2147 int ast_sip_thread_is_servant(void)
2149 uint32_t *servant_id;
2151 servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
2156 return *servant_id == SIP_SERVANT_ID;
2159 void *ast_sip_dict_get(void *ht, const char *key)
2161 unsigned int hval = 0;
2167 return pj_hash_get(ht, key, PJ_HASH_KEY_STRING, &hval);
2170 void *ast_sip_dict_set(pj_pool_t* pool, void *ht,
2171 const char *key, void *val)
2174 ht = pj_hash_create(pool, 11);
2177 pj_hash_set(pool, ht, key, PJ_HASH_KEY_STRING, 0, val);
2182 static pj_bool_t supplement_on_rx_request(pjsip_rx_data *rdata)
2184 struct ast_sip_supplement *supplement;
2186 if (pjsip_rdata_get_dlg(rdata)) {
2190 AST_RWLIST_RDLOCK(&supplements);
2191 AST_LIST_TRAVERSE(&supplements, supplement, next) {
2192 if (supplement->incoming_request && does_method_match(&rdata->msg_info.msg->line.req.method.name, supplement->method)) {
2193 supplement->incoming_request(ast_pjsip_rdata_get_endpoint(rdata), rdata);
2196 AST_RWLIST_UNLOCK(&supplements);
2201 int ast_sip_send_response(pjsip_response_addr *res_addr, pjsip_tx_data *tdata, struct ast_sip_endpoint *sip_endpoint)
2203 struct ast_sip_supplement *supplement;
2204 pjsip_cseq_hdr *cseq = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_CSEQ, NULL);
2205 struct ast_sip_contact *contact = ast_sip_mod_data_get(tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT);
2207 AST_RWLIST_RDLOCK(&supplements);
2208 AST_LIST_TRAVERSE(&supplements, supplement, next) {
2209 if (supplement->outgoing_response && does_method_match(&cseq->method.name, supplement->method)) {
2210 supplement->outgoing_response(sip_endpoint, contact, tdata);
2213 AST_RWLIST_UNLOCK(&supplements);
2215 ast_sip_mod_data_set(tdata->pool, tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT, NULL);
2216 ao2_cleanup(contact);
2218 return pjsip_endpt_send_response(ast_sip_get_pjsip_endpoint(), res_addr, tdata, NULL, NULL);
2221 int ast_sip_create_response(const pjsip_rx_data *rdata, int st_code,
2222 struct ast_sip_contact *contact, pjsip_tx_data **tdata)
2224 int res = pjsip_endpt_create_response(ast_sip_get_pjsip_endpoint(), rdata, st_code, NULL, tdata);
2227 ast_sip_mod_data_set((*tdata)->pool, (*tdata)->mod_data, supplement_module.id, MOD_DATA_CONTACT, ao2_bump(contact));
2233 static void remove_request_headers(pjsip_endpoint *endpt)
2235 const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt);
2236 pjsip_hdr *iter = request_headers->next;
2238 while (iter != request_headers) {
2239 pjsip_hdr *to_erase = iter;
2241 pj_list_erase(to_erase);
2245 static int load_module(void)
2247 /* The third parameter is just copied from
2248 * example code from PJLIB. This can be adjusted
2252 struct ast_threadpool_options options;
2254 if (pj_init() != PJ_SUCCESS) {
2255 return AST_MODULE_LOAD_DECLINE;
2258 if (pjlib_util_init() != PJ_SUCCESS) {
2260 return AST_MODULE_LOAD_DECLINE;
2263 pj_caching_pool_init(&caching_pool, NULL, 1024 * 1024);
2264 if (pjsip_endpt_create(&caching_pool.factory, "SIP", &ast_pjsip_endpoint) != PJ_SUCCESS) {
2265 ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n");
2266 pj_caching_pool_destroy(&caching_pool);
2267 return AST_MODULE_LOAD_DECLINE;
2270 /* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that,
2271 * we need to stop PJSIP from doing it automatically
2273 remove_request_headers(ast_pjsip_endpoint);
2275 memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL);
2277 ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n");
2278 pjsip_endpt_destroy(ast_pjsip_endpoint);
2279 ast_pjsip_endpoint = NULL;
2280 pj_caching_pool_destroy(&caching_pool);
2281 return AST_MODULE_LOAD_DECLINE;
2284 if (ast_sip_initialize_system()) {
2285 ast_log(LOG_ERROR, "Failed to initialize SIP 'system' configuration section. Aborting load\n");
2286 pj_pool_release(memory_pool);
2288 pjsip_endpt_destroy(ast_pjsip_endpoint);
2289 ast_pjsip_endpoint = NULL;
2290 pj_caching_pool_destroy(&caching_pool);
2291 return AST_MODULE_LOAD_DECLINE;
2294 sip_get_threadpool_options(&options);
2295 options.thread_start = sip_thread_start;
2296 sip_threadpool = ast_threadpool_create("SIP", NULL, &options);
2297 if (!sip_threadpool) {
2298 ast_log(LOG_ERROR, "Failed to create SIP threadpool. Aborting load\n");
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 pjsip_tsx_layer_init_module(ast_pjsip_endpoint);
2308 pjsip_ua_init_module(ast_pjsip_endpoint, NULL);
2310 monitor_continue = 1;
2311 status = pj_thread_create(memory_pool, "SIP", (pj_thread_proc *) &monitor_thread_exec,
2312 NULL, PJ_THREAD_DEFAULT_STACK_SIZE * 2, 0, &monitor_thread);
2313 if (status != PJ_SUCCESS) {
2314 ast_log(LOG_ERROR, "Failed to start SIP monitor thread. Aborting load\n");
2315 pj_pool_release(memory_pool);
2317 pjsip_endpt_destroy(ast_pjsip_endpoint);
2318 ast_pjsip_endpoint = NULL;
2319 pj_caching_pool_destroy(&caching_pool);
2320 return AST_MODULE_LOAD_DECLINE;
2323 ast_sip_initialize_global_headers();
2325 if (ast_res_pjsip_initialize_configuration(ast_module_info)) {
2326 ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
2327 ast_sip_destroy_global_headers();
2328 stop_monitor_thread();
2329 pj_pool_release(memory_pool);
2331 pjsip_endpt_destroy(ast_pjsip_endpoint);
2332 ast_pjsip_endpoint = NULL;
2333 pj_caching_pool_destroy(&caching_pool);
2334 return AST_MODULE_LOAD_DECLINE;
2337 if (ast_sip_initialize_distributor()) {
2338 ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n");
2339 ast_res_pjsip_destroy_configuration();
2340 ast_sip_destroy_global_headers();
2341 stop_monitor_thread();
2342 pj_pool_release(memory_pool);
2344 pjsip_endpt_destroy(ast_pjsip_endpoint);
2345 ast_pjsip_endpoint = NULL;
2346 pj_caching_pool_destroy(&caching_pool);
2347 return AST_MODULE_LOAD_DECLINE;
2350 if (ast_sip_register_service(&supplement_module)) {
2351 ast_log(LOG_ERROR, "Failed to initialize supplement hooks. Aborting load\n");
2352 ast_sip_destroy_distributor();
2353 ast_res_pjsip_destroy_configuration();
2354 ast_sip_destroy_global_headers();
2355 stop_monitor_thread();
2356 pj_pool_release(memory_pool);
2358 pjsip_endpt_destroy(ast_pjsip_endpoint);
2359 ast_pjsip_endpoint = NULL;
2360 pj_caching_pool_destroy(&caching_pool);
2361 return AST_MODULE_LOAD_DECLINE;
2364 if (ast_sip_initialize_outbound_authentication()) {
2365 ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n");
2366 ast_sip_unregister_service(&supplement_module);
2367 ast_sip_destroy_distributor();
2368 ast_res_pjsip_destroy_configuration();
2369 ast_sip_destroy_global_headers();
2370 stop_monitor_thread();
2371 pj_pool_release(memory_pool);
2373 pjsip_endpt_destroy(ast_pjsip_endpoint);
2374 ast_pjsip_endpoint = NULL;
2375 pj_caching_pool_destroy(&caching_pool);
2376 return AST_MODULE_LOAD_DECLINE;
2379 ast_res_pjsip_init_options_handling(0);
2381 ast_module_ref(ast_module_info->self);
2383 return AST_MODULE_LOAD_SUCCESS;
2386 static int reload_module(void)
2388 if (ast_res_pjsip_reload_configuration()) {
2389 return AST_MODULE_LOAD_DECLINE;
2391 ast_res_pjsip_init_options_handling(1);
2395 static int unload_module(void)
2397 /* This will never get called as this module can't be unloaded */
2401 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Basic SIP resource",
2402 .load = load_module,
2403 .unload = unload_module,
2404 .reload = reload_module,
2405 .load_pri = AST_MODPRI_CHANNEL_DEPEND - 5,
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