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="dtmfmode" 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="mailboxes">
248 <synopsis>Mailbox(es) to be associated with</synopsis>
250 <configOption name="mohsuggest" default="default">
251 <synopsis>Default Music On Hold class</synopsis>
253 <configOption name="outbound_auth">
254 <synopsis>Authentication object used for outbound requests</synopsis>
256 <configOption name="outbound_proxy">
257 <synopsis>Proxy through which to send requests</synopsis>
259 <configOption name="rewrite_contact">
260 <synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
262 On inbound SIP messages from this endpoint, the Contact header will be changed to have the
263 source IP address and port. This option does not affect outbound messages send to this
265 </para></description>
267 <configOption name="rtp_ipv6" default="no">
268 <synopsis>Allow use of IPv6 for RTP traffic</synopsis>
270 <configOption name="rtp_symmetric" default="no">
271 <synopsis>Enforce that RTP must be symmetric</synopsis>
273 <configOption name="send_diversion" default="yes">
274 <synopsis>Send the Diversion header, conveying the diversion
275 information to the called user agent</synopsis>
277 <configOption name="send_pai" default="no">
278 <synopsis>Send the P-Asserted-Identity header</synopsis>
280 <configOption name="send_rpid" default="no">
281 <synopsis>Send the Remote-Party-ID header</synopsis>
283 <configOption name="timers_min_se" default="90">
284 <synopsis>Minimum session timers expiration period</synopsis>
286 Minimium session timer expiration period. Time in seconds.
287 </para></description>
289 <configOption name="timers" default="yes">
290 <synopsis>Session timers for SIP packets</synopsis>
293 <enum name="forced" />
295 <enum name="required" />
300 <configOption name="timers_sess_expires" default="1800">
301 <synopsis>Maximum session timer expiration period</synopsis>
303 Maximium session timer expiration period. Time in seconds.
304 </para></description>
306 <configOption name="transport">
307 <synopsis>Desired transport configuration</synopsis>
309 This will set the desired transport configuration to send SIP data through.
311 <warning><para>Not specifying a transport will <emphasis>DEFAULT</emphasis>
312 to the first configured transport in <filename>pjsip.conf</filename> which is
313 valid for the URI we are trying to contact.
315 <warning><para>Transport configuration is not affected by reloads. In order to
316 change transports, a full Asterisk restart is required</para></warning>
319 <configOption name="trust_id_inbound" default="no">
320 <synopsis>Accept identification information received from this endpoint</synopsis>
321 <description><para>This option determines whether Asterisk will accept
322 identification from the endpoint from headers such as P-Asserted-Identity
323 or Remote-Party-ID header. This option applies both to calls originating from the
324 endpoint and calls originating from Asterisk. If <literal>no</literal>, the
325 configured Caller-ID from pjsip.conf will always be used as the identity for
326 the endpoint.</para></description>
328 <configOption name="trust_id_outbound" default="no">
329 <synopsis>Send private identification details to the endpoint.</synopsis>
330 <description><para>This option determines whether res_pjsip will send private
331 identification information to the endpoint. If <literal>no</literal>,
332 private Caller-ID information will not be forwarded to the endpoint.
333 "Private" in this case refers to any method of restricting identification.
334 Example: setting <replaceable>callerid_privacy</replaceable> to any
335 <literal>prohib</literal> variation.
336 Example: If <replaceable>trust_id_inbound</replaceable> is set to
337 <literal>yes</literal>, the presence of a <literal>Privacy: id</literal>
338 header in a SIP request or response would indicate the identification
339 provided in the request is private.</para></description>
341 <configOption name="type">
342 <synopsis>Must be of type 'endpoint'.</synopsis>
344 <configOption name="use_ptime" default="no">
345 <synopsis>Use Endpoint's requested packetisation interval</synopsis>
347 <configOption name="use_avpf" default="no">
348 <synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this
351 If set to <literal>yes</literal>, res_pjsip will use use the AVPF or SAVPF RTP
352 profile for all media offers on outbound calls and media updates and will
353 decline media offers not using the AVPF or SAVPF profile.
355 If set to <literal>no</literal>, res_pjsip will use use the AVP or SAVP RTP
356 profile for all media offers on outbound calls and media updates and will
357 decline media offers not using the AVP or SAVP profile.
358 </para></description>
360 <configOption name="media_encryption" default="no">
361 <synopsis>Determines whether res_pjsip will use and enforce usage of media encryption
362 for this endpoint.</synopsis>
365 <enum name="no"><para>
366 res_pjsip will offer no encryption and allow no encryption to be setup.
368 <enum name="sdes"><para>
369 res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP
370 transport should be used in conjunction with this option to prevent
371 exposure of media encryption keys.
373 <enum name="dtls"><para>
374 res_pjsip will offer DTLS-SRTP setup.
379 <configOption name="inband_progress" default="no">
380 <synopsis>Determines whether chan_pjsip will indicate ringing using inband
383 If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
384 when told to indicate ringing and will immediately start sending ringing
387 If set to <literal>no</literal>, chan_pjsip will send a 180 Ringing when told
388 to indicate ringing and will NOT send it as audio.
389 </para></description>
391 <configOption name="callgroup">
392 <synopsis>The numeric pickup groups for a channel.</synopsis>
394 Can be set to a comma separated list of numbers or ranges between the values
395 of 0-63 (maximum of 64 groups).
396 </para></description>
398 <configOption name="pickupgroup">
399 <synopsis>The numeric pickup groups that a channel can pickup.</synopsis>
401 Can be set to a comma separated list of numbers or ranges between the values
402 of 0-63 (maximum of 64 groups).
403 </para></description>
405 <configOption name="namedcallgroup">
406 <synopsis>The named pickup groups for a channel.</synopsis>
408 Can be set to a comma separated list of case sensitive strings limited by
409 supported line length.
410 </para></description>
412 <configOption name="namedpickupgroup">
413 <synopsis>The named pickup groups that a channel can pickup.</synopsis>
415 Can be set to a comma separated list of case sensitive strings limited by
416 supported line length.
417 </para></description>
419 <configOption name="devicestate_busy_at" default="0">
420 <synopsis>The number of in-use channels which will cause busy to be returned as device state</synopsis>
422 When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the
423 PJSIP channel driver will return busy as the device state instead of in use.
424 </para></description>
426 <configOption name="t38udptl" default="no">
427 <synopsis>Whether T.38 UDPTL support is enabled or not</synopsis>
429 If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted
431 </para></description>
433 <configOption name="t38udptl_ec" default="none">
434 <synopsis>T.38 UDPTL error correction method</synopsis>
437 <enum name="none"><para>
438 No error correction should be used.
440 <enum name="fec"><para>
441 Forward error correction should be used.
443 <enum name="redundancy"><para>
444 Redundacy error correction should be used.
449 <configOption name="t38udptl_maxdatagram" default="0">
450 <synopsis>T.38 UDPTL maximum datagram size</synopsis>
452 This option can be set to override the maximum datagram of a remote endpoint for broken
454 </para></description>
456 <configOption name="faxdetect" default="no">
457 <synopsis>Whether CNG tone detection is enabled</synopsis>
459 This option can be set to send the session to the fax extension when a CNG tone is
461 </para></description>
463 <configOption name="t38udptl_nat" default="no">
464 <synopsis>Whether NAT support is enabled on UDPTL sessions</synopsis>
466 When enabled the UDPTL stack will send UDPTL packets to the source address of
468 </para></description>
470 <configOption name="t38udptl_ipv6" default="no">
471 <synopsis>Whether IPv6 is used for UDPTL Sessions</synopsis>
473 When enabled the UDPTL stack will use IPv6.
474 </para></description>
476 <configOption name="tonezone">
477 <synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis>
479 <configOption name="language">
480 <synopsis>Set the default language to use for channels created for this endpoint.</synopsis>
482 <configOption name="one_touch_recording" default="no">
483 <synopsis>Determines whether one-touch recording is allowed for this endpoint.</synopsis>
485 <ref type="configOption">recordonfeature</ref>
486 <ref type="configOption">recordofffeature</ref>
489 <configOption name="recordonfeature" default="automixmon">
490 <synopsis>The feature to enact when one-touch recording is turned on.</synopsis>
492 <para>When an INFO request for one-touch recording arrives with a Record header set to "on", this
493 feature will be enabled for the channel. The feature designated here can be any built-in
494 or dynamic feature defined in features.conf.</para>
495 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
498 <ref type="configOption">one_touch_recording</ref>
499 <ref type="configOption">recordofffeature</ref>
502 <configOption name="recordofffeature" default="automixmon">
503 <synopsis>The feature to enact when one-touch recording is turned off.</synopsis>
505 <para>When an INFO request for one-touch recording arrives with a Record header set to "off", this
506 feature will be enabled for the channel. The feature designated here can be any built-in
507 or dynamic feature defined in features.conf.</para>
508 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
511 <ref type="configOption">one_touch_recording</ref>
512 <ref type="configOption">recordonfeature</ref>
515 <configOption name="rtpengine" default="asterisk">
516 <synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis>
518 <configOption name="allowtransfer" default="yes">
519 <synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis>
521 <configOption name="sdpowner" default="-">
522 <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
524 <configOption name="sdpsession" default="Asterisk">
525 <synopsis>String used for the SDP session (s=) line.</synopsis>
527 <configOption name="tos_audio">
528 <synopsis>DSCP TOS bits for audio streams</synopsis>
530 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
531 </para></description>
533 <configOption name="tos_video">
534 <synopsis>DSCP TOS bits for video streams</synopsis>
536 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
537 </para></description>
539 <configOption name="cos_audio">
540 <synopsis>Priority for audio streams</synopsis>
542 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
543 </para></description>
545 <configOption name="cos_video">
546 <synopsis>Priority for video streams</synopsis>
548 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
549 </para></description>
551 <configOption name="allowsubscribe" default="yes">
552 <synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis>
554 <configOption name="subminexpiry" default="60">
555 <synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis>
557 <configOption name="fromuser">
558 <synopsis>Username to use in From header for requests to this endpoint.</synopsis>
560 <configOption name="mwifromuser">
561 <synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis>
563 <configOption name="fromdomain">
564 <synopsis>Domain to user in From header for requests to this endpoint.</synopsis>
566 <configOption name="dtlsverify">
567 <synopsis>Verify that the provided peer certificate is valid</synopsis>
569 This option only applies if <replaceable>media_encryption</replaceable> is
570 set to <literal>dtls</literal>.
571 </para></description>
573 <configOption name="dtlsrekey">
574 <synopsis>Interval at which to renegotiate the TLS session and rekey the SRTP session</synopsis>
576 This option only applies if <replaceable>media_encryption</replaceable> is
577 set to <literal>dtls</literal>.
579 If this is not set or the value provided is 0 rekeying will be disabled.
580 </para></description>
582 <configOption name="dtlscertfile">
583 <synopsis>Path to certificate file to present to peer</synopsis>
585 This option only applies if <replaceable>media_encryption</replaceable> is
586 set to <literal>dtls</literal>.
587 </para></description>
589 <configOption name="dtlsprivatekey">
590 <synopsis>Path to private key for certificate file</synopsis>
592 This option only applies if <replaceable>media_encryption</replaceable> is
593 set to <literal>dtls</literal>.
594 </para></description>
596 <configOption name="dtlscipher">
597 <synopsis>Cipher to use for DTLS negotiation</synopsis>
599 This option only applies if <replaceable>media_encryption</replaceable> is
600 set to <literal>dtls</literal>.
602 Many options for acceptable ciphers. See link for more:
603 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
604 </para></description>
606 <configOption name="dtlscafile">
607 <synopsis>Path to certificate authority certificate</synopsis>
609 This option only applies if <replaceable>media_encryption</replaceable> is
610 set to <literal>dtls</literal>.
611 </para></description>
613 <configOption name="dtlscapath">
614 <synopsis>Path to a directory containing certificate authority certificates</synopsis>
616 This option only applies if <replaceable>media_encryption</replaceable> is
617 set to <literal>dtls</literal>.
618 </para></description>
620 <configOption name="dtlssetup">
621 <synopsis>Whether we are willing to accept connections, connect to the other party, or both.</synopsis>
624 This option only applies if <replaceable>media_encryption</replaceable> is
625 set to <literal>dtls</literal>.
628 <enum name="active"><para>
629 res_pjsip will make a connection to the peer.
631 <enum name="passive"><para>
632 res_pjsip will accept connections from the peer.
634 <enum name="actpass"><para>
635 res_pjsip will offer and accept connections from the peer.
640 <configOption name="srtp_tag_32">
641 <synopsis>Determines whether 32 byte tags should be used instead of 80 byte tags.</synopsis>
643 This option only applies if <replaceable>media_encryption</replaceable> is
644 set to <literal>sdes</literal> or <literal>dtls</literal>.
645 </para></description>
648 <configObject name="auth">
649 <synopsis>Authentication type</synopsis>
651 Authentication objects hold the authentication information for use
652 by other objects such as <literal>endpoints</literal> or <literal>registrations</literal>.
653 This also allows for multiple objects to use a single auth object. See
654 the <literal>auth_type</literal> config option for password style choices.
655 </para></description>
656 <configOption name="auth_type" default="userpass">
657 <synopsis>Authentication type</synopsis>
659 This option specifies which of the password style config options should be read
660 when trying to authenticate an endpoint inbound request. If set to <literal>userpass</literal>
661 then we'll read from the 'password' option. For <literal>md5</literal> we'll read
666 <enum name="userpass"/>
670 <configOption name="nonce_lifetime" default="32">
671 <synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
673 <configOption name="md5_cred">
674 <synopsis>MD5 Hash used for authentication.</synopsis>
675 <description><para>Only used when auth_type is <literal>md5</literal>.</para></description>
677 <configOption name="password">
678 <synopsis>PlainText password used for authentication.</synopsis>
679 <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
681 <configOption name="realm" default="asterisk">
682 <synopsis>SIP realm for endpoint</synopsis>
684 <configOption name="type">
685 <synopsis>Must be 'auth'</synopsis>
687 <configOption name="username">
688 <synopsis>Username to use for account</synopsis>
691 <configObject name="domain_alias">
692 <synopsis>Domain Alias</synopsis>
694 Signifies that a domain is an alias. If the domain on a session is
695 not found to match an AoR then this object is used to see if we have
696 an alias for the AoR to which the endpoint is binding. This objects
697 name as defined in configuration should be the domain alias and a
698 config option is provided to specify the domain to be aliased.
699 </para></description>
700 <configOption name="type">
701 <synopsis>Must be of type 'domain_alias'.</synopsis>
703 <configOption name="domain">
704 <synopsis>Domain to be aliased</synopsis>
707 <configObject name="transport">
708 <synopsis>SIP Transport</synopsis>
710 <emphasis>Transports</emphasis>
712 <para>There are different transports and protocol derivatives
713 supported by <literal>res_pjsip</literal>. They are in order of
714 preference: UDP, TCP, and WebSocket (WS).</para>
715 <note><para>Changes to transport configuration in pjsip.conf will only be
716 effected on a complete restart of Asterisk. A module reload
717 will not suffice.</para></note>
719 <configOption name="async_operations" default="1">
720 <synopsis>Number of simultaneous Asynchronous Operations</synopsis>
722 <configOption name="bind">
723 <synopsis>IP Address and optional port to bind to for this transport</synopsis>
725 <configOption name="ca_list_file">
726 <synopsis>File containing a list of certificates to read (TLS ONLY)</synopsis>
728 <configOption name="cert_file">
729 <synopsis>Certificate file for endpoint (TLS ONLY)</synopsis>
731 <configOption name="cipher">
732 <synopsis>Preferred Cryptography Cipher (TLS ONLY)</synopsis>
734 Many options for acceptable ciphers see link for more:
735 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
736 </para></description>
738 <configOption name="domain">
739 <synopsis>Domain the transport comes from</synopsis>
741 <configOption name="external_media_address">
742 <synopsis>External IP address to use in RTP handling</synopsis>
744 When a request or response is sent out, if the destination of the
745 message is outside the IP network defined in the option <literal>localnet</literal>,
746 and the media address in the SDP is within the localnet network, then the
747 media address in the SDP will be rewritten to the value defined for
748 <literal>external_media_address</literal>.
749 </para></description>
751 <configOption name="external_signaling_address">
752 <synopsis>External address for SIP signalling</synopsis>
754 <configOption name="external_signaling_port" default="0">
755 <synopsis>External port for SIP signalling</synopsis>
757 <configOption name="method">
758 <synopsis>Method of SSL transport (TLS ONLY)</synopsis>
761 <enum name="default" />
762 <enum name="unspecified" />
763 <enum name="tlsv1" />
764 <enum name="sslv2" />
765 <enum name="sslv3" />
766 <enum name="sslv23" />
770 <configOption name="localnet">
771 <synopsis>Network to consider local (used for NAT purposes).</synopsis>
772 <description><para>This must be in CIDR or dotted decimal format with the IP
773 and mask separated with a slash ('/').</para></description>
775 <configOption name="password">
776 <synopsis>Password required for transport</synopsis>
778 <configOption name="privkey_file">
779 <synopsis>Private key file (TLS ONLY)</synopsis>
781 <configOption name="protocol" default="udp">
782 <synopsis>Protocol to use for SIP traffic</synopsis>
793 <configOption name="require_client_cert" default="false">
794 <synopsis>Require client certificate (TLS ONLY)</synopsis>
796 <configOption name="type">
797 <synopsis>Must be of type 'transport'.</synopsis>
799 <configOption name="verify_client" default="false">
800 <synopsis>Require verification of client certificate (TLS ONLY)</synopsis>
802 <configOption name="verify_server" default="false">
803 <synopsis>Require verification of server certificate (TLS ONLY)</synopsis>
805 <configOption name="tos" default="false">
806 <synopsis>Enable TOS for the signalling sent over this transport</synopsis>
808 <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
809 for more information on this parameter.</para>
810 <note><para>This option does not apply to the <replaceable>ws</replaceable>
811 or the <replaceable>wss</replaceable> protocols.</para></note>
814 <configOption name="cos" default="false">
815 <synopsis>Enable COS for the signalling sent over this transport</synopsis>
817 <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
818 for more information on this parameter.</para>
819 <note><para>This option does not apply to the <replaceable>ws</replaceable>
820 or the <replaceable>wss</replaceable> protocols.</para></note>
824 <configObject name="contact">
825 <synopsis>A way of creating an aliased name to a SIP URI</synopsis>
827 Contacts are a way to hide SIP URIs from the dialplan directly.
828 They are also used to make a group of contactable parties when
829 in use with <literal>AoR</literal> lists.
830 </para></description>
831 <configOption name="type">
832 <synopsis>Must be of type 'contact'.</synopsis>
834 <configOption name="uri">
835 <synopsis>SIP URI to contact peer</synopsis>
837 <configOption name="expiration_time">
838 <synopsis>Time to keep alive a contact</synopsis>
840 Time to keep alive a contact. String style specification.
841 </para></description>
843 <configOption name="qualify_frequency" default="0">
844 <synopsis>Interval at which to qualify a contact</synopsis>
846 Interval between attempts to qualify the contact for reachability.
847 If <literal>0</literal> never qualify. Time in seconds.
848 </para></description>
851 <configObject name="aor">
852 <synopsis>The configuration for a location of an endpoint</synopsis>
854 An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
855 AoRs are specified, an endpoint will not be reachable by Asterisk.
856 Beyond that, an AoR has other uses within Asterisk, such as inbound
859 An <literal>AoR</literal> is a way to allow dialing a group
860 of <literal>Contacts</literal> that all use the same
861 <literal>endpoint</literal> for calls.
863 This can be used as another way of grouping a list of contacts to dial
864 rather than specifing them each directly when dialing via the dialplan.
865 This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
867 Registrations: For Asterisk to match an inbound registration to an endpoint,
868 the AoR object name must match the user portion of the SIP URI in the "To:"
869 header of the inbound SIP registration. That will usually be equivalent
870 to the "user name" set in your hard or soft phones configuration.
871 </para></description>
872 <configOption name="contact">
873 <synopsis>Permanent contacts assigned to AoR</synopsis>
875 Contacts specified will be called whenever referenced
876 by <literal>chan_pjsip</literal>.
878 Use a separate "contact=" entry for each contact required. Contacts
879 are specified using a SIP URI.
880 </para></description>
882 <configOption name="default_expiration" default="3600">
883 <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
885 <configOption name="mailboxes">
886 <synopsis>Mailbox(es) to be associated with</synopsis>
887 <description><para>This option applies when an external entity subscribes to an AoR
888 for message waiting indications. The mailboxes specified will be subscribed to.
889 More than one mailbox can be specified with a comma-delimited string.</para></description>
891 <configOption name="maximum_expiration" default="7200">
892 <synopsis>Maximum time to keep an AoR</synopsis>
894 Maximium time to keep a peer with explicit expiration. Time in seconds.
895 </para></description>
897 <configOption name="max_contacts" default="0">
898 <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
900 Maximum number of contacts that can associate with this AoR. This value does
901 not affect the number of contacts that can be added with the "contact" option.
902 It only limits contacts added through external interaction, such as
905 <note><para>This should be set to <literal>1</literal> and
906 <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
907 wish to stick with the older <literal>chan_sip</literal> behaviour.
911 <configOption name="minimum_expiration" default="60">
912 <synopsis>Minimum keep alive time for an AoR</synopsis>
914 Minimum time to keep a peer with an explict expiration. Time in seconds.
915 </para></description>
917 <configOption name="remove_existing" default="no">
918 <synopsis>Determines whether new contacts replace existing ones.</synopsis>
920 On receiving a new registration to the AoR should it remove
921 the existing contact that was registered against it?
923 <note><para>This should be set to <literal>yes</literal> and
924 <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
925 wish to stick with the older <literal>chan_sip</literal> behaviour.
929 <configOption name="type">
930 <synopsis>Must be of type 'aor'.</synopsis>
932 <configOption name="qualify_frequency" default="0">
933 <synopsis>Interval at which to qualify an AoR</synopsis>
935 Interval between attempts to qualify the AoR for reachability.
936 If <literal>0</literal> never qualify. Time in seconds.
937 </para></description>
939 <configOption name="authenticate_qualify" default="no">
940 <synopsis>Authenticates a qualify request if needed</synopsis>
942 If true and a qualify request receives a challenge or authenticate response
943 authentication is attempted before declaring the contact available.
944 </para></description>
947 <configObject name="system">
948 <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
950 The settings in this section are global. In addition to being global, the values will
951 not be re-evaluated when a reload is performed. This is because the values must be set
952 before the SIP stack is initialized. The only way to reset these values is to either
953 restart Asterisk, or unload res_pjsip.so and then load it again.
954 </para></description>
955 <configOption name="timert1" default="500">
956 <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
958 Timer T1 is the base for determining how long to wait before retransmitting
959 requests that receive no response when using an unreliable transport (e.g. UDP).
960 For more information on this timer, see RFC 3261, Section 17.1.1.1.
961 </para></description>
963 <configOption name="timerb" default="32000">
964 <synopsis>Set transaction timer B value (milliseconds).</synopsis>
966 Timer B determines the maximum amount of time to wait after sending an INVITE
967 request before terminating the transaction. It is recommended that this be set
968 to 64 * Timer T1, but it may be set higher if desired. For more information on
969 this timer, see RFC 3261, Section 17.1.1.1.
970 </para></description>
972 <configOption name="compactheaders" default="no">
973 <synopsis>Use the short forms of common SIP header names.</synopsis>
975 <configOption name="threadpool_initial_size" default="0">
976 <synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
978 <configOption name="threadpool_auto_increment" default="5">
979 <synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
981 <configOption name="threadpool_idle_timeout" default="60">
982 <synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
984 <configOption name="threadpool_max_size" default="0">
985 <synopsis>Maximum number of threads in the res_pjsip threadpool.
986 A value of 0 indicates no maximum.</synopsis>
988 <configOption name="type">
989 <synopsis>Must be of type 'system'.</synopsis>
992 <configObject name="global">
993 <synopsis>Options that apply globally to all SIP communications</synopsis>
995 The settings in this section are global. Unlike options in the <literal>system</literal>
996 section, these options can be refreshed by performing a reload.
997 </para></description>
998 <configOption name="maxforwards" default="70">
999 <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
1001 <configOption name="type">
1002 <synopsis>Must be of type 'global'.</synopsis>
1004 <configOption name="useragent" default="Asterisk <Asterisk Version>">
1005 <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
1010 <manager name="PJSIPQualify" language="en_US">
1012 Qualify a chan_pjsip endpoint.
1015 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1016 <parameter name="Endpoint" required="true">
1017 <para>The endpoint you want to qualify.</para>
1021 <para>Qualify a chan_pjsip endpoint.</para>
1027 static pjsip_endpoint *ast_pjsip_endpoint;
1029 static struct ast_threadpool *sip_threadpool;
1031 static int register_service(void *data)
1033 pjsip_module **module = data;
1034 if (!ast_pjsip_endpoint) {
1035 ast_log(LOG_ERROR, "There is no PJSIP endpoint. Unable to register services\n");
1038 if (pjsip_endpt_register_module(ast_pjsip_endpoint, *module) != PJ_SUCCESS) {
1039 ast_log(LOG_ERROR, "Unable to register module %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1042 ast_debug(1, "Registered SIP service %.*s (%p)\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name), *module);
1043 ast_module_ref(ast_module_info->self);
1047 int ast_sip_register_service(pjsip_module *module)
1049 return ast_sip_push_task_synchronous(NULL, register_service, &module);
1052 static int unregister_service(void *data)
1054 pjsip_module **module = data;
1055 ast_module_unref(ast_module_info->self);
1056 if (!ast_pjsip_endpoint) {
1059 pjsip_endpt_unregister_module(ast_pjsip_endpoint, *module);
1060 ast_debug(1, "Unregistered SIP service %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1064 void ast_sip_unregister_service(pjsip_module *module)
1066 ast_sip_push_task_synchronous(NULL, unregister_service, &module);
1069 static struct ast_sip_authenticator *registered_authenticator;
1071 int ast_sip_register_authenticator(struct ast_sip_authenticator *auth)
1073 if (registered_authenticator) {
1074 ast_log(LOG_WARNING, "Authenticator %p is already registered. Cannot register a new one\n", registered_authenticator);
1077 registered_authenticator = auth;
1078 ast_debug(1, "Registered SIP authenticator module %p\n", auth);
1079 ast_module_ref(ast_module_info->self);
1083 void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth)
1085 if (registered_authenticator != auth) {
1086 ast_log(LOG_WARNING, "Trying to unregister authenticator %p but authenticator %p registered\n",
1087 auth, registered_authenticator);
1090 registered_authenticator = NULL;
1091 ast_debug(1, "Unregistered SIP authenticator %p\n", auth);
1092 ast_module_unref(ast_module_info->self);
1095 int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1097 if (!registered_authenticator) {
1098 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is not required\n");
1102 return registered_authenticator->requires_authentication(endpoint, rdata);
1105 enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
1106 pjsip_rx_data *rdata, pjsip_tx_data *tdata)
1108 if (!registered_authenticator) {
1109 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is successful\n");
1112 return registered_authenticator->check_authentication(endpoint, rdata, tdata);
1115 static struct ast_sip_outbound_authenticator *registered_outbound_authenticator;
1117 int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1119 if (registered_outbound_authenticator) {
1120 ast_log(LOG_WARNING, "Outbound authenticator %p is already registered. Cannot register a new one\n", registered_outbound_authenticator);
1123 registered_outbound_authenticator = auth;
1124 ast_debug(1, "Registered SIP outbound authenticator module %p\n", auth);
1125 ast_module_ref(ast_module_info->self);
1129 void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1131 if (registered_outbound_authenticator != auth) {
1132 ast_log(LOG_WARNING, "Trying to unregister outbound authenticator %p but outbound authenticator %p registered\n",
1133 auth, registered_outbound_authenticator);
1136 registered_outbound_authenticator = NULL;
1137 ast_debug(1, "Unregistered SIP outbound authenticator %p\n", auth);
1138 ast_module_unref(ast_module_info->self);
1141 int ast_sip_create_request_with_auth(const struct ast_sip_auth_array *auths, pjsip_rx_data *challenge,
1142 pjsip_transaction *tsx, pjsip_tx_data **new_request)
1144 if (!registered_outbound_authenticator) {
1145 ast_log(LOG_WARNING, "No SIP outbound authenticator registered. Cannot respond to authentication challenge\n");
1148 return registered_outbound_authenticator->create_request_with_auth(auths, challenge, tsx, new_request);
1151 struct endpoint_identifier_list {
1152 struct ast_sip_endpoint_identifier *identifier;
1153 AST_RWLIST_ENTRY(endpoint_identifier_list) list;
1156 static AST_RWLIST_HEAD_STATIC(endpoint_identifiers, endpoint_identifier_list);
1158 int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1160 struct endpoint_identifier_list *id_list_item;
1161 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1163 id_list_item = ast_calloc(1, sizeof(*id_list_item));
1164 if (!id_list_item) {
1165 ast_log(LOG_ERROR, "Unabled to add endpoint identifier. Out of memory.\n");
1168 id_list_item->identifier = identifier;
1170 AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list);
1171 ast_debug(1, "Registered endpoint identifier %p\n", identifier);
1173 ast_module_ref(ast_module_info->self);
1177 void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1179 struct endpoint_identifier_list *iter;
1180 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1181 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) {
1182 if (iter->identifier == identifier) {
1183 AST_RWLIST_REMOVE_CURRENT(list);
1185 ast_debug(1, "Unregistered endpoint identifier %p\n", identifier);
1186 ast_module_unref(ast_module_info->self);
1190 AST_RWLIST_TRAVERSE_SAFE_END;
1193 struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata)
1195 struct endpoint_identifier_list *iter;
1196 struct ast_sip_endpoint *endpoint = NULL;
1197 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1198 AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) {
1199 ast_assert(iter->identifier->identify_endpoint != NULL);
1200 endpoint = iter->identifier->identify_endpoint(rdata);
1208 pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
1210 return ast_pjsip_endpoint;
1213 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)
1215 pj_str_t tmp, local_addr;
1217 pjsip_sip_uri *sip_uri;
1218 pjsip_transport_type_e type = PJSIP_TRANSPORT_UNSPECIFIED;
1220 char uuid_str[AST_UUID_STR_LEN];
1222 if (ast_strlen_zero(user)) {
1223 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1227 user = ast_uuid_to_str(uuid, uuid_str, sizeof(uuid_str));
1230 /* Parse the provided target URI so we can determine what transport it will end up using */
1231 pj_strdup_with_null(pool, &tmp, target);
1233 if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) ||
1234 (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
1238 sip_uri = pjsip_uri_get_uri(uri);
1240 /* Determine the transport type to use */
1241 if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
1242 type = PJSIP_TRANSPORT_TLS;
1243 } else if (!sip_uri->transport_param.slen) {
1244 type = PJSIP_TRANSPORT_UDP;
1246 type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
1249 if (type == PJSIP_TRANSPORT_UNSPECIFIED) {
1253 /* If the host is IPv6 turn the transport into an IPv6 version */
1254 if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1255 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1258 if (!ast_strlen_zero(domain)) {
1259 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1260 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1262 (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1265 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1266 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1270 /* Get the local bound address for the transport that will be used when communicating with the provided URI */
1271 if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), pool, type, selector,
1272 &local_addr, &local_port) != PJ_SUCCESS) {
1276 /* If IPv6 was specified in the transport, set the proper type */
1277 if (pj_strchr(&local_addr, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1278 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1281 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1282 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1283 "<%s:%s@%s%.*s%s:%d%s%s>",
1284 (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1286 (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1287 (int)local_addr.slen,
1289 (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1291 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1292 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1297 static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
1299 RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
1300 const char *transport_name = endpoint->transport;
1302 if (ast_strlen_zero(transport_name)) {
1306 transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name);
1308 if (!transport || !transport->state) {
1309 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport '%s' for endpoint '%s'\n",
1310 transport_name, ast_sorcery_object_get_id(endpoint));
1314 if (transport->state->transport) {
1315 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1316 selector->u.transport = transport->state->transport;
1317 } else if (transport->state->factory) {
1318 selector->type = PJSIP_TPSELECTOR_LISTENER;
1319 selector->u.listener = transport->state->factory;
1327 static int sip_get_tpselector_from_uri(const char *uri, pjsip_tpselector *selector)
1329 RAII_VAR(struct ast_sip_contact_transport *, contact_transport, NULL, ao2_cleanup);
1331 contact_transport = ast_sip_location_retrieve_contact_transport_by_uri(uri);
1333 if (!contact_transport) {
1337 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1338 selector->u.transport = contact_transport->transport;
1343 pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
1345 pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri;
1346 pjsip_dialog *dlg = NULL;
1347 const char *outbound_proxy = endpoint->outbound_proxy;
1348 pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1349 static const pj_str_t HCONTACT = { "Contact", 7 };
1351 pj_cstr(&remote_uri, uri);
1353 if (pjsip_dlg_create_uac(pjsip_ua_instance(), &local_uri, NULL, &remote_uri, NULL, &dlg) != PJ_SUCCESS) {
1357 if (sip_get_tpselector_from_uri(uri, &selector) && sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1358 pjsip_dlg_terminate(dlg);
1362 if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
1363 pjsip_dlg_terminate(dlg);
1367 /* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
1368 pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
1369 dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
1370 dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
1372 /* If a request user has been specified and we are permitted to change it, do so */
1373 if (!ast_strlen_zero(request_user) && (PJSIP_URI_SCHEME_IS_SIP(dlg->target) || PJSIP_URI_SCHEME_IS_SIPS(dlg->target))) {
1374 pjsip_sip_uri *target = pjsip_uri_get_uri(dlg->target);
1375 pj_strdup2(dlg->pool, &target->user, request_user);
1378 /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
1381 pjsip_dlg_set_transport(dlg, &selector);
1383 if (!ast_strlen_zero(outbound_proxy)) {
1384 pjsip_route_hdr route_set, *route;
1385 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1388 pj_list_init(&route_set);
1390 pj_strdup2_with_null(dlg->pool, &tmp, outbound_proxy);
1391 if (!(route = pjsip_parse_hdr(dlg->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1392 pjsip_dlg_terminate(dlg);
1395 pj_list_push_back(&route_set, route);
1397 pjsip_dlg_set_route_set(dlg, &route_set);
1405 pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1409 pjsip_transport_type_e type = rdata->tp_info.transport->key.type;
1412 contact.ptr = pj_pool_alloc(rdata->tp_info.pool, PJSIP_MAX_URL_SIZE);
1413 contact.slen = pj_ansi_snprintf(contact.ptr, PJSIP_MAX_URL_SIZE,
1414 "<%s:%s%.*s%s:%d%s%s>",
1415 (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1416 (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1417 (int)rdata->tp_info.transport->local_name.host.slen,
1418 rdata->tp_info.transport->local_name.host.ptr,
1419 (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1420 rdata->tp_info.transport->local_name.port,
1421 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1422 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1424 status = pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, &contact, &dlg);
1425 if (status != PJ_SUCCESS) {
1426 char err[PJ_ERR_MSG_SIZE];
1428 pjsip_strerror(status, err, sizeof(err));
1429 ast_log(LOG_ERROR, "Could not create dialog with endpoint %s. %s\n",
1430 ast_sorcery_object_get_id(endpoint), err);
1437 /* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
1438 static const pjsip_method info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
1439 static const pjsip_method message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
1443 const pjsip_method *pmethod;
1445 { "INVITE", &pjsip_invite_method },
1446 { "CANCEL", &pjsip_cancel_method },
1447 { "ACK", &pjsip_ack_method },
1448 { "BYE", &pjsip_bye_method },
1449 { "REGISTER", &pjsip_register_method },
1450 { "OPTIONS", &pjsip_options_method },
1451 { "SUBSCRIBE", &pjsip_subscribe_method },
1452 { "NOTIFY", &pjsip_notify_method },
1453 { "PUBLISH", &pjsip_publish_method },
1454 { "INFO", &info_method },
1455 { "MESSAGE", &message_method },
1458 static const pjsip_method *get_pjsip_method(const char *method)
1461 for (i = 0; i < ARRAY_LEN(methods); ++i) {
1462 if (!strcmp(method, methods[i].method)) {
1463 return methods[i].pmethod;
1469 static int create_in_dialog_request(const pjsip_method *method, struct pjsip_dialog *dlg, pjsip_tx_data **tdata)
1471 if (pjsip_dlg_create_request(dlg, method, -1, tdata) != PJ_SUCCESS) {
1472 ast_log(LOG_WARNING, "Unable to create in-dialog request.\n");
1479 static int create_out_of_dialog_request(const pjsip_method *method, struct ast_sip_endpoint *endpoint,
1480 const char *uri, pjsip_tx_data **tdata)
1482 RAII_VAR(struct ast_sip_contact *, contact, NULL, ao2_cleanup);
1483 pj_str_t remote_uri;
1486 pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1488 if (ast_strlen_zero(uri)) {
1490 ast_log(LOG_ERROR, "An endpoint and/or uri must be specified\n");
1494 contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
1495 if (!contact || ast_strlen_zero(contact->uri)) {
1496 ast_log(LOG_ERROR, "Unable to retrieve contact for endpoint %s\n",
1497 ast_sorcery_object_get_id(endpoint));
1501 pj_cstr(&remote_uri, contact->uri);
1503 pj_cstr(&remote_uri, uri);
1507 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1508 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
1509 ast_sorcery_object_get_id(endpoint));
1514 pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
1517 ast_log(LOG_ERROR, "Unable to create PJLIB memory pool\n");
1521 if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL,
1522 endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
1523 ast_log(LOG_ERROR, "Unable to create From header for %.*s request to endpoint %s\n",
1524 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1525 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1529 if (pjsip_endpt_create_request(ast_sip_get_pjsip_endpoint(), method, &remote_uri,
1530 &from, &remote_uri, &from, NULL, -1, NULL, tdata) != PJ_SUCCESS) {
1531 ast_log(LOG_ERROR, "Unable to create outbound %.*s request to endpoint %s\n",
1532 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1533 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1537 /* We can release this pool since request creation copied all the necessary
1538 * data into the outbound request's pool
1540 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1544 int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
1545 struct ast_sip_endpoint *endpoint, const char *uri,
1546 pjsip_tx_data **tdata)
1548 const pjsip_method *pmethod = get_pjsip_method(method);
1551 ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
1556 return create_in_dialog_request(pmethod, dlg, tdata);
1558 return create_out_of_dialog_request(pmethod, endpoint, uri, tdata);
1562 static int send_in_dialog_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg)
1564 if (pjsip_dlg_send_request(dlg, tdata, -1, NULL) != PJ_SUCCESS) {
1565 ast_log(LOG_WARNING, "Unable to send in-dialog request.\n");
1571 static void send_request_cb(void *token, pjsip_event *e)
1573 RAII_VAR(struct ast_sip_endpoint *, endpoint, token, ao2_cleanup);
1574 pjsip_transaction *tsx = e->body.tsx_state.tsx;
1575 pjsip_rx_data *challenge = e->body.tsx_state.src.rdata;
1576 pjsip_tx_data *tdata;
1578 if (tsx->status_code != 401 && tsx->status_code != 407) {
1582 if (!ast_sip_create_request_with_auth(&endpoint->outbound_auths, challenge, tsx, &tdata)) {
1583 pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, NULL, NULL);
1587 static int send_out_of_dialog_request(pjsip_tx_data *tdata, struct ast_sip_endpoint *endpoint)
1589 ao2_ref(endpoint, +1);
1590 if (pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, endpoint, send_request_cb) != PJ_SUCCESS) {
1591 ast_log(LOG_ERROR, "Error attempting to send outbound %.*s request to endpoint %s\n",
1592 (int) pj_strlen(&tdata->msg->line.req.method.name),
1593 pj_strbuf(&tdata->msg->line.req.method.name),
1594 ast_sorcery_object_get_id(endpoint));
1595 ao2_ref(endpoint, -1);
1602 int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint)
1604 ast_assert(tdata->msg->type == PJSIP_REQUEST_MSG);
1607 return send_in_dialog_request(tdata, dlg);
1609 return send_out_of_dialog_request(tdata, endpoint);
1613 int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value)
1617 pjsip_generic_string_hdr *hdr;
1619 pj_cstr(&hdr_name, name);
1620 pj_cstr(&hdr_value, value);
1622 hdr = pjsip_generic_string_hdr_create(tdata->pool, &hdr_name, &hdr_value);
1624 pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) hdr);
1628 static pjsip_msg_body *ast_body_to_pjsip_body(pj_pool_t *pool, const struct ast_sip_body *body)
1634 pj_cstr(&type, body->type);
1635 pj_cstr(&subtype, body->subtype);
1636 pj_cstr(&body_text, body->body_text);
1638 return pjsip_msg_body_create(pool, &type, &subtype, &body_text);
1641 int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body)
1643 pjsip_msg_body *pjsip_body = ast_body_to_pjsip_body(tdata->pool, body);
1644 tdata->msg->body = pjsip_body;
1648 int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies)
1651 /* NULL for type and subtype automatically creates "multipart/mixed" */
1652 pjsip_msg_body *body = pjsip_multipart_create(tdata->pool, NULL, NULL);
1654 for (i = 0; i < num_bodies; ++i) {
1655 pjsip_multipart_part *part = pjsip_multipart_create_part(tdata->pool);
1656 part->body = ast_body_to_pjsip_body(tdata->pool, bodies[i]);
1657 pjsip_multipart_add_part(tdata->pool, body, part);
1660 tdata->msg->body = body;
1664 int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text)
1666 size_t combined_size = strlen(body_text) + tdata->msg->body->len;
1667 struct ast_str *body_buffer = ast_str_alloca(combined_size);
1669 ast_str_set(&body_buffer, 0, "%.*s%s", (int) tdata->msg->body->len, (char *) tdata->msg->body->data, body_text);
1671 tdata->msg->body->data = pj_pool_alloc(tdata->pool, combined_size);
1672 pj_memcpy(tdata->msg->body->data, ast_str_buffer(body_buffer), combined_size);
1673 tdata->msg->body->len = combined_size;
1678 struct ast_taskprocessor *ast_sip_create_serializer(void)
1680 struct ast_taskprocessor *serializer;
1681 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1682 char name[AST_UUID_STR_LEN];
1688 ast_uuid_to_str(uuid, name, sizeof(name));
1690 serializer = ast_threadpool_serializer(name, sip_threadpool);
1697 int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1700 return ast_taskprocessor_push(serializer, sip_task, task_data);
1702 return ast_threadpool_push(sip_threadpool, sip_task, task_data);
1706 struct sync_task_data {
1711 int (*task)(void *);
1715 static int sync_task(void *data)
1717 struct sync_task_data *std = data;
1718 std->fail = std->task(std->task_data);
1720 ast_mutex_lock(&std->lock);
1722 ast_cond_signal(&std->cond);
1723 ast_mutex_unlock(&std->lock);
1727 int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1729 /* This method is an onion */
1730 struct sync_task_data std;
1731 ast_mutex_init(&std.lock);
1732 ast_cond_init(&std.cond, NULL);
1733 std.fail = std.complete = 0;
1734 std.task = sip_task;
1735 std.task_data = task_data;
1738 if (ast_taskprocessor_push(serializer, sync_task, &std)) {
1742 if (ast_threadpool_push(sip_threadpool, sync_task, &std)) {
1747 ast_mutex_lock(&std.lock);
1748 while (!std.complete) {
1749 ast_cond_wait(&std.cond, &std.lock);
1751 ast_mutex_unlock(&std.lock);
1753 ast_mutex_destroy(&std.lock);
1754 ast_cond_destroy(&std.cond);
1758 void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
1760 size_t chars_to_copy = MIN(size - 1, pj_strlen(src));
1761 memcpy(dest, pj_strbuf(src), chars_to_copy);
1762 dest[chars_to_copy] = '\0';
1765 int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype)
1767 pjsip_media_type compare;
1769 if (!content_type) {
1773 pjsip_media_type_init2(&compare, type, subtype);
1775 return pjsip_media_type_cmp(content_type, &compare, 0) ? 0 : -1;
1778 pj_caching_pool caching_pool;
1779 pj_pool_t *memory_pool;
1780 pj_thread_t *monitor_thread;
1781 static int monitor_continue;
1783 static void *monitor_thread_exec(void *endpt)
1785 while (monitor_continue) {
1786 const pj_time_val delay = {0, 10};
1787 pjsip_endpt_handle_events(ast_pjsip_endpoint, &delay);
1792 static void stop_monitor_thread(void)
1794 monitor_continue = 0;
1795 pj_thread_join(monitor_thread);
1798 AST_THREADSTORAGE(pj_thread_storage);
1799 AST_THREADSTORAGE(servant_id_storage);
1800 #define SIP_SERVANT_ID 0x5E2F1D
1802 static void sip_thread_start(void)
1804 pj_thread_desc *desc;
1805 pj_thread_t *thread;
1806 uint32_t *servant_id;
1808 servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1810 ast_log(LOG_ERROR, "Could not set SIP servant ID in thread-local storage.\n");
1813 *servant_id = SIP_SERVANT_ID;
1815 desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc));
1817 ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n");
1820 pj_bzero(*desc, sizeof(*desc));
1822 if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) {
1823 ast_log(LOG_ERROR, "Couldn't register thread with PJLIB.\n");
1827 int ast_sip_thread_is_servant(void)
1829 uint32_t *servant_id;
1831 servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1836 return *servant_id == SIP_SERVANT_ID;
1839 void *ast_sip_dict_get(void *ht, const char *key)
1847 return pj_hash_get(ht, key, PJ_HASH_KEY_STRING, &hval);
1850 void *ast_sip_dict_set(pj_pool_t* pool, void *ht,
1851 const char *key, void *val)
1854 ht = pj_hash_create(pool, 11);
1857 pj_hash_set(pool, ht, key, PJ_HASH_KEY_STRING, 0, val);
1862 static void remove_request_headers(pjsip_endpoint *endpt)
1864 const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt);
1865 pjsip_hdr *iter = request_headers->next;
1867 while (iter != request_headers) {
1868 pjsip_hdr *to_erase = iter;
1870 pj_list_erase(to_erase);
1874 static int load_module(void)
1876 /* The third parameter is just copied from
1877 * example code from PJLIB. This can be adjusted
1881 struct ast_threadpool_options options;
1883 if (pj_init() != PJ_SUCCESS) {
1884 return AST_MODULE_LOAD_DECLINE;
1887 if (pjlib_util_init() != PJ_SUCCESS) {
1889 return AST_MODULE_LOAD_DECLINE;
1892 pj_caching_pool_init(&caching_pool, NULL, 1024 * 1024);
1893 if (pjsip_endpt_create(&caching_pool.factory, "SIP", &ast_pjsip_endpoint) != PJ_SUCCESS) {
1894 ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n");
1895 pj_caching_pool_destroy(&caching_pool);
1896 return AST_MODULE_LOAD_DECLINE;
1899 /* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that,
1900 * we need to stop PJSIP from doing it automatically
1902 remove_request_headers(ast_pjsip_endpoint);
1904 memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL);
1906 ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n");
1907 pjsip_endpt_destroy(ast_pjsip_endpoint);
1908 ast_pjsip_endpoint = NULL;
1909 pj_caching_pool_destroy(&caching_pool);
1910 return AST_MODULE_LOAD_DECLINE;
1913 if (ast_sip_initialize_system()) {
1914 ast_log(LOG_ERROR, "Failed to initialize SIP system configuration. Aborting load\n");
1915 pj_pool_release(memory_pool);
1917 pjsip_endpt_destroy(ast_pjsip_endpoint);
1918 ast_pjsip_endpoint = NULL;
1919 pj_caching_pool_destroy(&caching_pool);
1920 return AST_MODULE_LOAD_DECLINE;
1923 sip_get_threadpool_options(&options);
1924 options.thread_start = sip_thread_start;
1925 sip_threadpool = ast_threadpool_create("SIP", NULL, &options);
1926 if (!sip_threadpool) {
1927 ast_log(LOG_ERROR, "Failed to create SIP threadpool. Aborting load\n");
1928 pj_pool_release(memory_pool);
1930 pjsip_endpt_destroy(ast_pjsip_endpoint);
1931 ast_pjsip_endpoint = NULL;
1932 pj_caching_pool_destroy(&caching_pool);
1933 return AST_MODULE_LOAD_DECLINE;
1936 pjsip_tsx_layer_init_module(ast_pjsip_endpoint);
1937 pjsip_ua_init_module(ast_pjsip_endpoint, NULL);
1939 monitor_continue = 1;
1940 status = pj_thread_create(memory_pool, "SIP", (pj_thread_proc *) &monitor_thread_exec,
1941 NULL, PJ_THREAD_DEFAULT_STACK_SIZE * 2, 0, &monitor_thread);
1942 if (status != PJ_SUCCESS) {
1943 ast_log(LOG_ERROR, "Failed to start SIP monitor thread. Aborting load\n");
1944 pj_pool_release(memory_pool);
1946 pjsip_endpt_destroy(ast_pjsip_endpoint);
1947 ast_pjsip_endpoint = NULL;
1948 pj_caching_pool_destroy(&caching_pool);
1949 return AST_MODULE_LOAD_DECLINE;
1952 ast_sip_initialize_global_headers();
1954 if (ast_res_pjsip_initialize_configuration()) {
1955 ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
1956 ast_sip_destroy_global_headers();
1957 stop_monitor_thread();
1958 pj_pool_release(memory_pool);
1960 pjsip_endpt_destroy(ast_pjsip_endpoint);
1961 ast_pjsip_endpoint = NULL;
1962 pj_caching_pool_destroy(&caching_pool);
1963 return AST_MODULE_LOAD_DECLINE;
1966 if (ast_sip_initialize_distributor()) {
1967 ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n");
1968 ast_res_pjsip_destroy_configuration();
1969 ast_sip_destroy_global_headers();
1970 stop_monitor_thread();
1971 pj_pool_release(memory_pool);
1973 pjsip_endpt_destroy(ast_pjsip_endpoint);
1974 ast_pjsip_endpoint = NULL;
1975 pj_caching_pool_destroy(&caching_pool);
1976 return AST_MODULE_LOAD_DECLINE;
1979 if (ast_sip_initialize_outbound_authentication()) {
1980 ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n");
1981 ast_sip_destroy_distributor();
1982 ast_res_pjsip_destroy_configuration();
1983 ast_sip_destroy_global_headers();
1984 stop_monitor_thread();
1985 pj_pool_release(memory_pool);
1987 pjsip_endpt_destroy(ast_pjsip_endpoint);
1988 ast_pjsip_endpoint = NULL;
1989 pj_caching_pool_destroy(&caching_pool);
1990 return AST_MODULE_LOAD_DECLINE;
1993 ast_res_pjsip_init_options_handling(0);
1995 ast_res_pjsip_init_contact_transports();
1997 ast_module_ref(ast_module_info->self);
1999 return AST_MODULE_LOAD_SUCCESS;
2002 static int reload_module(void)
2004 if (ast_res_pjsip_reload_configuration()) {
2005 return AST_MODULE_LOAD_DECLINE;
2007 ast_res_pjsip_init_options_handling(1);
2011 static int unload_module(void)
2013 /* This will never get called as this module can't be unloaded */
2017 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Basic SIP resource",
2018 .load = load_module,
2019 .unload = unload_module,
2020 .reload = reload_module,
2021 .load_pri = AST_MODPRI_CHANNEL_DEPEND - 5,
projects / asterisk/asterisk.git / blob