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="external_media_address">
212 <synopsis>IP used for External Media handling</synopsis>
214 <configOption name="force_rport" default="yes">
215 <synopsis>Force use of return port</synopsis>
217 <configOption name="ice_support" default="no">
218 <synopsis>Enable the ICE mechanism to help traverse NAT</synopsis>
220 <configOption name="identify_by" default="username,location">
221 <synopsis>Way(s) for Endpoint to be identified</synopsis>
223 An endpoint can be identified in multiple ways. Currently, the only supported
224 option is <literal>username</literal>, which matches the endpoint based on the
225 username in the From header.
227 <note><para>Endpoints can also be identified by IP address; however, that method
228 of identification is not handled by this configuration option. See the documentation
229 for the <literal>identify</literal> configuration section for more details on that
230 method of endpoint identification. If this option is set to <literal>username</literal>
231 and an <literal>identify</literal> configuration section exists for the endpoint, then
232 the endpoint can be identified in multiple ways.</para></note>
234 <enum name="username" />
238 <configOption name="mailboxes">
239 <synopsis>Mailbox(es) to be associated with</synopsis>
241 <configOption name="mohsuggest" default="default">
242 <synopsis>Default Music On Hold class</synopsis>
244 <configOption name="outbound_auth">
245 <synopsis>Authentication object used for outbound requests</synopsis>
247 <configOption name="outbound_proxy">
248 <synopsis>Proxy through which to send requests</synopsis>
250 <configOption name="rewrite_contact">
251 <synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
253 <configOption name="rtp_ipv6" default="no">
254 <synopsis>Allow use of IPv6 for RTP traffic</synopsis>
256 <configOption name="rtp_symmetric" default="no">
257 <synopsis>Enforce that RTP must be symmetric</synopsis>
259 <configOption name="send_diversion" default="yes">
260 <synopsis>Send the Diversion header, conveying the diversion
261 information to the called user agent</synopsis>
263 <configOption name="send_pai" default="no">
264 <synopsis>Send the P-Asserted-Identity header</synopsis>
266 <configOption name="send_rpid" default="no">
267 <synopsis>Send the Remote-Party-ID header</synopsis>
269 <configOption name="timers_min_se" default="90">
270 <synopsis>Minimum session timers expiration period</synopsis>
272 Minimium session timer expiration period. Time in seconds.
273 </para></description>
275 <configOption name="timers" default="yes">
276 <synopsis>Session timers for SIP packets</synopsis>
279 <enum name="forced" />
281 <enum name="required" />
286 <configOption name="timers_sess_expires" default="1800">
287 <synopsis>Maximum session timer expiration period</synopsis>
289 Maximium session timer expiration period. Time in seconds.
290 </para></description>
292 <configOption name="transport">
293 <synopsis>Desired transport configuration</synopsis>
295 This will set the desired transport configuration to send SIP data through.
297 <warning><para>Not specifying a transport will <emphasis>DEFAULT</emphasis>
298 to the first configured transport in <filename>pjsip.conf</filename> which is
299 valid for the URI we are trying to contact.
301 <warning><para>Transport configuration is not affected by reloads. In order to
302 change transports, a full Asterisk restart is required</para></warning>
305 <configOption name="trust_id_inbound" default="no">
306 <synopsis>Accept identification information received from this endpoint</synopsis>
307 <description><para>This option determines whether Asterisk will accept
308 identification from the endpoint from headers such as P-Asserted-Identity
309 or Remote-Party-ID header. This option applies both to calls originating from the
310 endpoint and calls originating from Asterisk. If <literal>no</literal>, the
311 configured Caller-ID from pjsip.conf will always be used as the identity for
312 the endpoint.</para></description>
314 <configOption name="trust_id_outbound" default="no">
315 <synopsis>Send private identification details to the endpoint.</synopsis>
316 <description><para>This option determines whether res_pjsip will send private
317 identification information to the endpoint. If <literal>no</literal>,
318 private Caller-ID information will not be forwarded to the endpoint.
319 "Private" in this case refers to any method of restricting identification.
320 Example: setting <replaceable>callerid_privacy</replaceable> to any
321 <literal>prohib</literal> variation.
322 Example: If <replaceable>trust_id_inbound</replaceable> is set to
323 <literal>yes</literal>, the presence of a <literal>Privacy: id</literal>
324 header in a SIP request or response would indicate the identification
325 provided in the request is private.</para></description>
327 <configOption name="type">
328 <synopsis>Must be of type 'endpoint'.</synopsis>
330 <configOption name="use_ptime" default="no">
331 <synopsis>Use Endpoint's requested packetisation interval</synopsis>
333 <configOption name="use_avpf" default="no">
334 <synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this
337 If set to <literal>yes</literal>, res_pjsip will use use the AVPF or SAVPF RTP
338 profile for all media offers on outbound calls and media updates and will
339 decline media offers not using the AVPF or SAVPF profile.
341 If set to <literal>no</literal>, res_pjsip will use use the AVP or SAVP RTP
342 profile for all media offers on outbound calls and media updates and will
343 decline media offers not using the AVP or SAVP profile.
344 </para></description>
346 <configOption name="media_encryption" default="no">
347 <synopsis>Determines whether res_pjsip will use and enforce usage of media encryption
348 for this endpoint.</synopsis>
351 <enum name="no"><para>
352 res_pjsip will offer no encryption and allow no encryption to be setup.
354 <enum name="sdes"><para>
355 res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP
356 transport should be used in conjunction with this option to prevent
357 exposure of media encryption keys.
359 <enum name="dtls"><para>
360 res_pjsip will offer DTLS-SRTP setup.
365 <configOption name="inband_progress" default="no">
366 <synopsis>Determines whether chan_pjsip will indicate ringing using inband
369 If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
370 when told to indicate ringing and will immediately start sending ringing
373 If set to <literal>no</literal>, chan_pjsip will send a 180 Ringing when told
374 to indicate ringing and will NOT send it as audio.
375 </para></description>
377 <configOption name="callgroup">
378 <synopsis>The numeric pickup groups for a channel.</synopsis>
380 Can be set to a comma separated list of numbers or ranges between the values
381 of 0-63 (maximum of 64 groups).
382 </para></description>
384 <configOption name="pickupgroup">
385 <synopsis>The numeric pickup groups that a channel can pickup.</synopsis>
387 Can be set to a comma separated list of numbers or ranges between the values
388 of 0-63 (maximum of 64 groups).
389 </para></description>
391 <configOption name="namedcallgroup">
392 <synopsis>The named pickup groups for a channel.</synopsis>
394 Can be set to a comma separated list of case sensitive strings limited by
395 supported line length.
396 </para></description>
398 <configOption name="namedpickupgroup">
399 <synopsis>The named pickup groups that a channel can pickup.</synopsis>
401 Can be set to a comma separated list of case sensitive strings limited by
402 supported line length.
403 </para></description>
405 <configOption name="devicestate_busy_at" default="0">
406 <synopsis>The number of in-use channels which will cause busy to be returned as device state</synopsis>
408 When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the
409 PJSIP channel driver will return busy as the device state instead of in use.
410 </para></description>
412 <configOption name="t38udptl" default="no">
413 <synopsis>Whether T.38 UDPTL support is enabled or not</synopsis>
415 If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted
417 </para></description>
419 <configOption name="t38udptl_ec" default="none">
420 <synopsis>T.38 UDPTL error correction method</synopsis>
423 <enum name="none"><para>
424 No error correction should be used.
426 <enum name="fec"><para>
427 Forward error correction should be used.
429 <enum name="redundancy"><para>
430 Redundacy error correction should be used.
435 <configOption name="t38udptl_maxdatagram" default="0">
436 <synopsis>T.38 UDPTL maximum datagram size</synopsis>
438 This option can be set to override the maximum datagram of a remote endpoint for broken
440 </para></description>
442 <configOption name="faxdetect" default="no">
443 <synopsis>Whether CNG tone detection is enabled</synopsis>
445 This option can be set to send the session to the fax extension when a CNG tone is
447 </para></description>
449 <configOption name="t38udptl_nat" default="no">
450 <synopsis>Whether NAT support is enabled on UDPTL sessions</synopsis>
452 When enabled the UDPTL stack will send UDPTL packets to the source address of
454 </para></description>
456 <configOption name="t38udptl_ipv6" default="no">
457 <synopsis>Whether IPv6 is used for UDPTL Sessions</synopsis>
459 When enabled the UDPTL stack will use IPv6.
460 </para></description>
462 <configOption name="tonezone">
463 <synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis>
465 <configOption name="language">
466 <synopsis>Set the default language to use for channels created for this endpoint.</synopsis>
468 <configOption name="one_touch_recording" default="no">
469 <synopsis>Determines whether one-touch recording is allowed for this endpoint.</synopsis>
471 <ref type="configOption">recordonfeature</ref>
472 <ref type="configOption">recordofffeature</ref>
475 <configOption name="recordonfeature" default="automixmon">
476 <synopsis>The feature to enact when one-touch recording is turned on.</synopsis>
478 <para>When an INFO request for one-touch recording arrives with a Record header set to "on", this
479 feature will be enabled for the channel. The feature designated here can be any built-in
480 or dynamic feature defined in features.conf.</para>
481 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
484 <ref type="configOption">one_touch_recording</ref>
485 <ref type="configOption">recordofffeature</ref>
488 <configOption name="recordofffeature" default="automixmon">
489 <synopsis>The feature to enact when one-touch recording is turned off.</synopsis>
491 <para>When an INFO request for one-touch recording arrives with a Record header set to "off", this
492 feature will be enabled for the channel. The feature designated here can be any built-in
493 or dynamic feature defined in features.conf.</para>
494 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
497 <ref type="configOption">one_touch_recording</ref>
498 <ref type="configOption">recordonfeature</ref>
501 <configOption name="rtpengine" default="asterisk">
502 <synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis>
504 <configOption name="allowtransfer" default="yes">
505 <synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis>
507 <configOption name="sdpowner" default="-">
508 <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
510 <configOption name="sdpsession" default="Asterisk">
511 <synopsis>String used for the SDP session (s=) line.</synopsis>
513 <configOption name="tos_audio">
514 <synopsis>DSCP TOS bits for audio streams</synopsis>
516 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
517 </para></description>
519 <configOption name="tos_video">
520 <synopsis>DSCP TOS bits for video streams</synopsis>
522 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
523 </para></description>
525 <configOption name="cos_audio">
526 <synopsis>Priority for audio streams</synopsis>
528 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
529 </para></description>
531 <configOption name="cos_video">
532 <synopsis>Priority for video streams</synopsis>
534 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
535 </para></description>
537 <configOption name="allowsubscribe" default="yes">
538 <synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis>
540 <configOption name="subminexpiry" default="60">
541 <synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis>
543 <configOption name="fromuser">
544 <synopsis>Username to use in From header for requests to this endpoint.</synopsis>
546 <configOption name="mwifromuser">
547 <synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis>
549 <configOption name="fromdomain">
550 <synopsis>Domain to user in From header for requests to this endpoint.</synopsis>
552 <configOption name="dtlsverify">
553 <synopsis>Verify that the provided peer certificate is valid</synopsis>
555 This option only applies if <replaceable>media_encryption</replaceable> is
556 set to <literal>dtls</literal>.
557 </para></description>
559 <configOption name="dtlsrekey">
560 <synopsis>Interval at which to renegotiate the TLS session and rekey the SRTP session</synopsis>
562 This option only applies if <replaceable>media_encryption</replaceable> is
563 set to <literal>dtls</literal>.
565 If this is not set or the value provided is 0 rekeying will be disabled.
566 </para></description>
568 <configOption name="dtlscertfile">
569 <synopsis>Path to certificate file to present to peer</synopsis>
571 This option only applies if <replaceable>media_encryption</replaceable> is
572 set to <literal>dtls</literal>.
573 </para></description>
575 <configOption name="dtlsprivatekey">
576 <synopsis>Path to private key for certificate file</synopsis>
578 This option only applies if <replaceable>media_encryption</replaceable> is
579 set to <literal>dtls</literal>.
580 </para></description>
582 <configOption name="dtlscipher">
583 <synopsis>Cipher to use for DTLS negotiation</synopsis>
585 This option only applies if <replaceable>media_encryption</replaceable> is
586 set to <literal>dtls</literal>.
588 Many options for acceptable ciphers. See link for more:
589 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
590 </para></description>
592 <configOption name="dtlscafile">
593 <synopsis>Path to certificate authority certificate</synopsis>
595 This option only applies if <replaceable>media_encryption</replaceable> is
596 set to <literal>dtls</literal>.
597 </para></description>
599 <configOption name="dtlscapath">
600 <synopsis>Path to a directory containing certificate authority certificates</synopsis>
602 This option only applies if <replaceable>media_encryption</replaceable> is
603 set to <literal>dtls</literal>.
604 </para></description>
606 <configOption name="dtlssetup">
607 <synopsis>Whether we are willing to accept connections, connect to the other party, or both.</synopsis>
610 This option only applies if <replaceable>media_encryption</replaceable> is
611 set to <literal>dtls</literal>.
614 <enum name="active"><para>
615 res_pjsip will make a connection to the peer.
617 <enum name="passive"><para>
618 res_pjsip will accept connections from the peer.
620 <enum name="actpass"><para>
621 res_pjsip will offer and accept connections from the peer.
626 <configOption name="srtp_tag_32">
627 <synopsis>Determines whether 32 byte tags should be used instead of 80 byte tags.</synopsis>
629 This option only applies if <replaceable>media_encryption</replaceable> is
630 set to <literal>sdes</literal> or <literal>dtls</literal>.
631 </para></description>
634 <configObject name="auth">
635 <synopsis>Authentication type</synopsis>
637 Authentication objects hold the authentication information for use
638 by other objects such as <literal>endpoints</literal> or <literal>registrations</literal>.
639 This also allows for multiple objects to use a single auth object. See
640 the <literal>auth_type</literal> config option for password style choices.
641 </para></description>
642 <configOption name="auth_type" default="userpass">
643 <synopsis>Authentication type</synopsis>
645 This option specifies which of the password style config options should be read
646 when trying to authenticate an endpoint inbound request. If set to <literal>userpass</literal>
647 then we'll read from the 'password' option. For <literal>md5</literal> we'll read
652 <enum name="userpass"/>
656 <configOption name="nonce_lifetime" default="32">
657 <synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
659 <configOption name="md5_cred">
660 <synopsis>MD5 Hash used for authentication.</synopsis>
661 <description><para>Only used when auth_type is <literal>md5</literal>.</para></description>
663 <configOption name="password">
664 <synopsis>PlainText password used for authentication.</synopsis>
665 <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
667 <configOption name="realm" default="asterisk">
668 <synopsis>SIP realm for endpoint</synopsis>
670 <configOption name="type">
671 <synopsis>Must be 'auth'</synopsis>
673 <configOption name="username">
674 <synopsis>Username to use for account</synopsis>
677 <configObject name="domain_alias">
678 <synopsis>Domain Alias</synopsis>
680 Signifies that a domain is an alias. If the domain on a session is
681 not found to match an AoR then this object is used to see if we have
682 an alias for the AoR to which the endpoint is binding. This objects
683 name as defined in configuration should be the domain alias and a
684 config option is provided to specify the domain to be aliased.
685 </para></description>
686 <configOption name="type">
687 <synopsis>Must be of type 'domain_alias'.</synopsis>
689 <configOption name="domain">
690 <synopsis>Domain to be aliased</synopsis>
693 <configObject name="transport">
694 <synopsis>SIP Transport</synopsis>
696 <emphasis>Transports</emphasis>
698 <para>There are different transports and protocol derivatives
699 supported by <literal>res_pjsip</literal>. They are in order of
700 preference: UDP, TCP, and WebSocket (WS).</para>
701 <note><para>Changes to transport configuration in pjsip.conf will only be
702 effected on a complete restart of Asterisk. A module reload
703 will not suffice.</para></note>
705 <configOption name="async_operations" default="1">
706 <synopsis>Number of simultaneous Asynchronous Operations</synopsis>
708 <configOption name="bind">
709 <synopsis>IP Address and optional port to bind to for this transport</synopsis>
711 <configOption name="ca_list_file">
712 <synopsis>File containing a list of certificates to read (TLS ONLY)</synopsis>
714 <configOption name="cert_file">
715 <synopsis>Certificate file for endpoint (TLS ONLY)</synopsis>
717 <configOption name="cipher">
718 <synopsis>Preferred Cryptography Cipher (TLS ONLY)</synopsis>
720 Many options for acceptable ciphers see link for more:
721 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
722 </para></description>
724 <configOption name="domain">
725 <synopsis>Domain the transport comes from</synopsis>
727 <configOption name="external_media_address">
728 <synopsis>External Address to use in RTP handling</synopsis>
730 <configOption name="external_signaling_address">
731 <synopsis>External address for SIP signalling</synopsis>
733 <configOption name="external_signaling_port" default="0">
734 <synopsis>External port for SIP signalling</synopsis>
736 <configOption name="method">
737 <synopsis>Method of SSL transport (TLS ONLY)</synopsis>
740 <enum name="default" />
741 <enum name="unspecified" />
742 <enum name="tlsv1" />
743 <enum name="sslv2" />
744 <enum name="sslv3" />
745 <enum name="sslv23" />
749 <configOption name="localnet">
750 <synopsis>Network to consider local (used for NAT purposes).</synopsis>
751 <description><para>This must be in CIDR or dotted decimal format with the IP
752 and mask separated with a slash ('/').</para></description>
754 <configOption name="password">
755 <synopsis>Password required for transport</synopsis>
757 <configOption name="privkey_file">
758 <synopsis>Private key file (TLS ONLY)</synopsis>
760 <configOption name="protocol" default="udp">
761 <synopsis>Protocol to use for SIP traffic</synopsis>
772 <configOption name="require_client_cert" default="false">
773 <synopsis>Require client certificate (TLS ONLY)</synopsis>
775 <configOption name="type">
776 <synopsis>Must be of type 'transport'.</synopsis>
778 <configOption name="verify_client" default="false">
779 <synopsis>Require verification of client certificate (TLS ONLY)</synopsis>
781 <configOption name="verify_server" default="false">
782 <synopsis>Require verification of server certificate (TLS ONLY)</synopsis>
784 <configOption name="tos" default="false">
785 <synopsis>Enable TOS for the signalling sent over this transport</synopsis>
787 <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
788 for more information on this parameter.</para>
789 <note><para>This option does not apply to the <replaceable>ws</replaceable>
790 or the <replaceable>wss</replaceable> protocols.</para></note>
793 <configOption name="cos" default="false">
794 <synopsis>Enable COS for the signalling sent over this transport</synopsis>
796 <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
797 for more information on this parameter.</para>
798 <note><para>This option does not apply to the <replaceable>ws</replaceable>
799 or the <replaceable>wss</replaceable> protocols.</para></note>
803 <configObject name="contact">
804 <synopsis>A way of creating an aliased name to a SIP URI</synopsis>
806 Contacts are a way to hide SIP URIs from the dialplan directly.
807 They are also used to make a group of contactable parties when
808 in use with <literal>AoR</literal> lists.
809 </para></description>
810 <configOption name="type">
811 <synopsis>Must be of type 'contact'.</synopsis>
813 <configOption name="uri">
814 <synopsis>SIP URI to contact peer</synopsis>
816 <configOption name="expiration_time">
817 <synopsis>Time to keep alive a contact</synopsis>
819 Time to keep alive a contact. String style specification.
820 </para></description>
822 <configOption name="qualify_frequency" default="0">
823 <synopsis>Interval at which to qualify a contact</synopsis>
825 Interval between attempts to qualify the contact for reachability.
826 If <literal>0</literal> never qualify. Time in seconds.
827 </para></description>
830 <configObject name="aor">
831 <synopsis>The configuration for a location of an endpoint</synopsis>
833 An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
834 AoRs are specified, an endpoint will not be reachable by Asterisk.
835 Beyond that, an AoR has other uses within Asterisk, such as inbound
838 An <literal>AoR</literal> is a way to allow dialing a group
839 of <literal>Contacts</literal> that all use the same
840 <literal>endpoint</literal> for calls.
842 This can be used as another way of grouping a list of contacts to dial
843 rather than specifing them each directly when dialing via the dialplan.
844 This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
846 Registrations: For Asterisk to match an inbound registration to an endpoint,
847 the AoR object name must match the user portion of the SIP URI in the "To:"
848 header of the inbound SIP registration. That will usually be equivalent
849 to the "user name" set in your hard or soft phones configuration.
850 </para></description>
851 <configOption name="contact">
852 <synopsis>Permanent contacts assigned to AoR</synopsis>
854 Contacts specified will be called whenever referenced
855 by <literal>chan_pjsip</literal>.
857 Use a separate "contact=" entry for each contact required. Contacts
858 are specified using a SIP URI.
859 </para></description>
861 <configOption name="default_expiration" default="3600">
862 <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
864 <configOption name="mailboxes">
865 <synopsis>Mailbox(es) to be associated with</synopsis>
866 <description><para>This option applies when an external entity subscribes to an AoR
867 for message waiting indications. The mailboxes specified will be subscribed to.
868 More than one mailbox can be specified with a comma-delimited string.</para></description>
870 <configOption name="maximum_expiration" default="7200">
871 <synopsis>Maximum time to keep an AoR</synopsis>
873 Maximium time to keep a peer with explicit expiration. Time in seconds.
874 </para></description>
876 <configOption name="max_contacts" default="0">
877 <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
879 Maximum number of contacts that can associate with this AoR. This value does
880 not affect the number of contacts that can be added with the "contact" option.
881 It only limits contacts added through external interaction, such as
884 <note><para>This should be set to <literal>1</literal> and
885 <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
886 wish to stick with the older <literal>chan_sip</literal> behaviour.
890 <configOption name="minimum_expiration" default="60">
891 <synopsis>Minimum keep alive time for an AoR</synopsis>
893 Minimum time to keep a peer with an explict expiration. Time in seconds.
894 </para></description>
896 <configOption name="remove_existing" default="no">
897 <synopsis>Determines whether new contacts replace existing ones.</synopsis>
899 On receiving a new registration to the AoR should it remove
900 the existing contact that was registered against it?
902 <note><para>This should be set to <literal>yes</literal> and
903 <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
904 wish to stick with the older <literal>chan_sip</literal> behaviour.
908 <configOption name="type">
909 <synopsis>Must be of type 'aor'.</synopsis>
911 <configOption name="qualify_frequency" default="0">
912 <synopsis>Interval at which to qualify an AoR</synopsis>
914 Interval between attempts to qualify the AoR for reachability.
915 If <literal>0</literal> never qualify. Time in seconds.
916 </para></description>
918 <configOption name="authenticate_qualify" default="no">
919 <synopsis>Authenticates a qualify request if needed</synopsis>
921 If true and a qualify request receives a challenge or authenticate response
922 authentication is attempted before declaring the contact available.
923 </para></description>
926 <configObject name="system">
927 <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
929 The settings in this section are global. In addition to being global, the values will
930 not be re-evaluated when a reload is performed. This is because the values must be set
931 before the SIP stack is initialized. The only way to reset these values is to either
932 restart Asterisk, or unload res_pjsip.so and then load it again.
933 </para></description>
934 <configOption name="timert1" default="500">
935 <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
937 Timer T1 is the base for determining how long to wait before retransmitting
938 requests that receive no response when using an unreliable transport (e.g. UDP).
939 For more information on this timer, see RFC 3261, Section 17.1.1.1.
940 </para></description>
942 <configOption name="timerb" default="32000">
943 <synopsis>Set transaction timer B value (milliseconds).</synopsis>
945 Timer B determines the maximum amount of time to wait after sending an INVITE
946 request before terminating the transaction. It is recommended that this be set
947 to 64 * Timer T1, but it may be set higher if desired. For more information on
948 this timer, see RFC 3261, Section 17.1.1.1.
949 </para></description>
951 <configOption name="compactheaders" default="no">
952 <synopsis>Use the short forms of common SIP header names.</synopsis>
954 <configOption name="threadpool_initial_size" default="0">
955 <synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
957 <configOption name="threadpool_auto_increment" default="5">
958 <synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
960 <configOption name="threadpool_idle_timeout" default="60">
961 <synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
963 <configOption name="threadpool_max_size" default="0">
964 <synopsis>Maximum number of threads in the res_pjsip threadpool.
965 A value of 0 indicates no maximum.</synopsis>
967 <configOption name="type">
968 <synopsis>Must be of type 'system'.</synopsis>
971 <configObject name="global">
972 <synopsis>Options that apply globally to all SIP communications</synopsis>
974 The settings in this section are global. Unlike options in the <literal>system</literal>
975 section, these options can be refreshed by performing a reload.
976 </para></description>
977 <configOption name="maxforwards" default="70">
978 <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
980 <configOption name="type">
981 <synopsis>Must be of type 'global'.</synopsis>
983 <configOption name="useragent" default="Asterisk <Asterisk Version>">
984 <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
989 <manager name="PJSIPQualify" language="en_US">
991 Qualify a chan_pjsip endpoint.
994 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
995 <parameter name="Endpoint" required="true">
996 <para>The endpoint you want to qualify.</para>
1000 <para>Qualify a chan_pjsip endpoint.</para>
1006 static pjsip_endpoint *ast_pjsip_endpoint;
1008 static struct ast_threadpool *sip_threadpool;
1010 static int register_service(void *data)
1012 pjsip_module **module = data;
1013 if (!ast_pjsip_endpoint) {
1014 ast_log(LOG_ERROR, "There is no PJSIP endpoint. Unable to register services\n");
1017 if (pjsip_endpt_register_module(ast_pjsip_endpoint, *module) != PJ_SUCCESS) {
1018 ast_log(LOG_ERROR, "Unable to register module %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1021 ast_debug(1, "Registered SIP service %.*s (%p)\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name), *module);
1022 ast_module_ref(ast_module_info->self);
1026 int ast_sip_register_service(pjsip_module *module)
1028 return ast_sip_push_task_synchronous(NULL, register_service, &module);
1031 static int unregister_service(void *data)
1033 pjsip_module **module = data;
1034 ast_module_unref(ast_module_info->self);
1035 if (!ast_pjsip_endpoint) {
1038 pjsip_endpt_unregister_module(ast_pjsip_endpoint, *module);
1039 ast_debug(1, "Unregistered SIP service %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1043 void ast_sip_unregister_service(pjsip_module *module)
1045 ast_sip_push_task_synchronous(NULL, unregister_service, &module);
1048 static struct ast_sip_authenticator *registered_authenticator;
1050 int ast_sip_register_authenticator(struct ast_sip_authenticator *auth)
1052 if (registered_authenticator) {
1053 ast_log(LOG_WARNING, "Authenticator %p is already registered. Cannot register a new one\n", registered_authenticator);
1056 registered_authenticator = auth;
1057 ast_debug(1, "Registered SIP authenticator module %p\n", auth);
1058 ast_module_ref(ast_module_info->self);
1062 void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth)
1064 if (registered_authenticator != auth) {
1065 ast_log(LOG_WARNING, "Trying to unregister authenticator %p but authenticator %p registered\n",
1066 auth, registered_authenticator);
1069 registered_authenticator = NULL;
1070 ast_debug(1, "Unregistered SIP authenticator %p\n", auth);
1071 ast_module_unref(ast_module_info->self);
1074 int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1076 if (!registered_authenticator) {
1077 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is not required\n");
1081 return registered_authenticator->requires_authentication(endpoint, rdata);
1084 enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
1085 pjsip_rx_data *rdata, pjsip_tx_data *tdata)
1087 if (!registered_authenticator) {
1088 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is successful\n");
1091 return registered_authenticator->check_authentication(endpoint, rdata, tdata);
1094 static struct ast_sip_outbound_authenticator *registered_outbound_authenticator;
1096 int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1098 if (registered_outbound_authenticator) {
1099 ast_log(LOG_WARNING, "Outbound authenticator %p is already registered. Cannot register a new one\n", registered_outbound_authenticator);
1102 registered_outbound_authenticator = auth;
1103 ast_debug(1, "Registered SIP outbound authenticator module %p\n", auth);
1104 ast_module_ref(ast_module_info->self);
1108 void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1110 if (registered_outbound_authenticator != auth) {
1111 ast_log(LOG_WARNING, "Trying to unregister outbound authenticator %p but outbound authenticator %p registered\n",
1112 auth, registered_outbound_authenticator);
1115 registered_outbound_authenticator = NULL;
1116 ast_debug(1, "Unregistered SIP outbound authenticator %p\n", auth);
1117 ast_module_unref(ast_module_info->self);
1120 int ast_sip_create_request_with_auth(const struct ast_sip_auth_array *auths, pjsip_rx_data *challenge,
1121 pjsip_transaction *tsx, pjsip_tx_data **new_request)
1123 if (!registered_outbound_authenticator) {
1124 ast_log(LOG_WARNING, "No SIP outbound authenticator registered. Cannot respond to authentication challenge\n");
1127 return registered_outbound_authenticator->create_request_with_auth(auths, challenge, tsx, new_request);
1130 struct endpoint_identifier_list {
1131 struct ast_sip_endpoint_identifier *identifier;
1132 AST_RWLIST_ENTRY(endpoint_identifier_list) list;
1135 static AST_RWLIST_HEAD_STATIC(endpoint_identifiers, endpoint_identifier_list);
1137 int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1139 struct endpoint_identifier_list *id_list_item;
1140 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1142 id_list_item = ast_calloc(1, sizeof(*id_list_item));
1143 if (!id_list_item) {
1144 ast_log(LOG_ERROR, "Unabled to add endpoint identifier. Out of memory.\n");
1147 id_list_item->identifier = identifier;
1149 AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list);
1150 ast_debug(1, "Registered endpoint identifier %p\n", identifier);
1152 ast_module_ref(ast_module_info->self);
1156 void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1158 struct endpoint_identifier_list *iter;
1159 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1160 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) {
1161 if (iter->identifier == identifier) {
1162 AST_RWLIST_REMOVE_CURRENT(list);
1164 ast_debug(1, "Unregistered endpoint identifier %p\n", identifier);
1165 ast_module_unref(ast_module_info->self);
1169 AST_RWLIST_TRAVERSE_SAFE_END;
1172 struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata)
1174 struct endpoint_identifier_list *iter;
1175 struct ast_sip_endpoint *endpoint = NULL;
1176 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1177 AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) {
1178 ast_assert(iter->identifier->identify_endpoint != NULL);
1179 endpoint = iter->identifier->identify_endpoint(rdata);
1187 pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
1189 return ast_pjsip_endpoint;
1192 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)
1194 pj_str_t tmp, local_addr;
1196 pjsip_sip_uri *sip_uri;
1197 pjsip_transport_type_e type = PJSIP_TRANSPORT_UNSPECIFIED;
1199 char uuid_str[AST_UUID_STR_LEN];
1201 if (ast_strlen_zero(user)) {
1202 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1206 user = ast_uuid_to_str(uuid, uuid_str, sizeof(uuid_str));
1209 /* Parse the provided target URI so we can determine what transport it will end up using */
1210 pj_strdup_with_null(pool, &tmp, target);
1212 if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) ||
1213 (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
1217 sip_uri = pjsip_uri_get_uri(uri);
1219 /* Determine the transport type to use */
1220 if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
1221 type = PJSIP_TRANSPORT_TLS;
1222 } else if (!sip_uri->transport_param.slen) {
1223 type = PJSIP_TRANSPORT_UDP;
1225 type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
1228 if (type == PJSIP_TRANSPORT_UNSPECIFIED) {
1232 /* If the host is IPv6 turn the transport into an IPv6 version */
1233 if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1234 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1237 if (!ast_strlen_zero(domain)) {
1238 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1239 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1241 (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1244 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1245 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1249 /* Get the local bound address for the transport that will be used when communicating with the provided URI */
1250 if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), pool, type, selector,
1251 &local_addr, &local_port) != PJ_SUCCESS) {
1255 /* If IPv6 was specified in the transport, set the proper type */
1256 if (pj_strchr(&local_addr, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1257 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1260 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1261 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1262 "<%s:%s@%s%.*s%s:%d%s%s>",
1263 (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1265 (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1266 (int)local_addr.slen,
1268 (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1270 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1271 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1276 static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
1278 RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
1279 const char *transport_name = endpoint->transport;
1281 if (ast_strlen_zero(transport_name)) {
1285 transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name);
1287 if (!transport || !transport->state) {
1291 if (transport->state->transport) {
1292 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1293 selector->u.transport = transport->state->transport;
1294 } else if (transport->state->factory) {
1295 selector->type = PJSIP_TPSELECTOR_LISTENER;
1296 selector->u.listener = transport->state->factory;
1304 static int sip_get_tpselector_from_uri(const char *uri, pjsip_tpselector *selector)
1306 RAII_VAR(struct ast_sip_contact_transport *, contact_transport, NULL, ao2_cleanup);
1308 contact_transport = ast_sip_location_retrieve_contact_transport_by_uri(uri);
1310 if (!contact_transport) {
1314 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1315 selector->u.transport = contact_transport->transport;
1320 pjsip_dialog *ast_sip_create_dialog(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
1322 pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri;
1323 pjsip_dialog *dlg = NULL;
1324 const char *outbound_proxy = endpoint->outbound_proxy;
1325 pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1326 static const pj_str_t HCONTACT = { "Contact", 7 };
1328 pj_cstr(&remote_uri, uri);
1330 if (pjsip_dlg_create_uac(pjsip_ua_instance(), &local_uri, NULL, &remote_uri, NULL, &dlg) != PJ_SUCCESS) {
1334 if (sip_get_tpselector_from_uri(uri, &selector) && sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1335 pjsip_dlg_terminate(dlg);
1339 if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
1340 pjsip_dlg_terminate(dlg);
1344 /* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
1345 pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
1346 dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
1347 dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
1349 /* If a request user has been specified and we are permitted to change it, do so */
1350 if (!ast_strlen_zero(request_user) && (PJSIP_URI_SCHEME_IS_SIP(dlg->target) || PJSIP_URI_SCHEME_IS_SIPS(dlg->target))) {
1351 pjsip_sip_uri *target = pjsip_uri_get_uri(dlg->target);
1352 pj_strdup2(dlg->pool, &target->user, request_user);
1355 /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
1358 pjsip_dlg_set_transport(dlg, &selector);
1360 if (!ast_strlen_zero(outbound_proxy)) {
1361 pjsip_route_hdr route_set, *route;
1362 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1365 pj_list_init(&route_set);
1367 pj_strdup2_with_null(dlg->pool, &tmp, outbound_proxy);
1368 if (!(route = pjsip_parse_hdr(dlg->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1369 pjsip_dlg_terminate(dlg);
1372 pj_list_push_back(&route_set, route);
1374 pjsip_dlg_set_route_set(dlg, &route_set);
1382 /* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
1383 const pjsip_method pjsip_info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
1384 const pjsip_method pjsip_message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
1388 const pjsip_method *pmethod;
1390 { "INVITE", &pjsip_invite_method },
1391 { "CANCEL", &pjsip_cancel_method },
1392 { "ACK", &pjsip_ack_method },
1393 { "BYE", &pjsip_bye_method },
1394 { "REGISTER", &pjsip_register_method },
1395 { "OPTIONS", &pjsip_options_method },
1396 { "SUBSCRIBE", &pjsip_subscribe_method },
1397 { "NOTIFY", &pjsip_notify_method },
1398 { "PUBLISH", &pjsip_publish_method },
1399 { "INFO", &pjsip_info_method },
1400 { "MESSAGE", &pjsip_message_method },
1403 static const pjsip_method *get_pjsip_method(const char *method)
1406 for (i = 0; i < ARRAY_LEN(methods); ++i) {
1407 if (!strcmp(method, methods[i].method)) {
1408 return methods[i].pmethod;
1414 static int create_in_dialog_request(const pjsip_method *method, struct pjsip_dialog *dlg, pjsip_tx_data **tdata)
1416 if (pjsip_dlg_create_request(dlg, method, -1, tdata) != PJ_SUCCESS) {
1417 ast_log(LOG_WARNING, "Unable to create in-dialog request.\n");
1424 static int create_out_of_dialog_request(const pjsip_method *method, struct ast_sip_endpoint *endpoint,
1425 const char *uri, pjsip_tx_data **tdata)
1427 RAII_VAR(struct ast_sip_contact *, contact, NULL, ao2_cleanup);
1428 pj_str_t remote_uri;
1431 pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1433 if (ast_strlen_zero(uri)) {
1435 ast_log(LOG_ERROR, "An endpoint and/or uri must be specified\n");
1439 contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
1440 if (!contact || ast_strlen_zero(contact->uri)) {
1441 ast_log(LOG_ERROR, "Unable to retrieve contact for endpoint %s\n",
1442 ast_sorcery_object_get_id(endpoint));
1446 pj_cstr(&remote_uri, contact->uri);
1448 pj_cstr(&remote_uri, uri);
1452 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1453 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
1454 ast_sorcery_object_get_id(endpoint));
1459 pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
1462 ast_log(LOG_ERROR, "Unable to create PJLIB memory pool\n");
1466 if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL,
1467 endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
1468 ast_log(LOG_ERROR, "Unable to create From header for %.*s request to endpoint %s\n",
1469 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1470 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1474 if (pjsip_endpt_create_request(ast_sip_get_pjsip_endpoint(), method, &remote_uri,
1475 &from, &remote_uri, &from, NULL, -1, NULL, tdata) != PJ_SUCCESS) {
1476 ast_log(LOG_ERROR, "Unable to create outbound %.*s request to endpoint %s\n",
1477 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1478 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1482 /* We can release this pool since request creation copied all the necessary
1483 * data into the outbound request's pool
1485 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1489 int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
1490 struct ast_sip_endpoint *endpoint, const char *uri,
1491 pjsip_tx_data **tdata)
1493 const pjsip_method *pmethod = get_pjsip_method(method);
1496 ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
1501 return create_in_dialog_request(pmethod, dlg, tdata);
1503 return create_out_of_dialog_request(pmethod, endpoint, uri, tdata);
1507 static int send_in_dialog_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg)
1509 if (pjsip_dlg_send_request(dlg, tdata, -1, NULL) != PJ_SUCCESS) {
1510 ast_log(LOG_WARNING, "Unable to send in-dialog request.\n");
1516 static void send_request_cb(void *token, pjsip_event *e)
1518 RAII_VAR(struct ast_sip_endpoint *, endpoint, token, ao2_cleanup);
1519 pjsip_transaction *tsx = e->body.tsx_state.tsx;
1520 pjsip_rx_data *challenge = e->body.tsx_state.src.rdata;
1521 pjsip_tx_data *tdata;
1523 if (tsx->status_code != 401 && tsx->status_code != 407) {
1527 if (!ast_sip_create_request_with_auth(&endpoint->outbound_auths, challenge, tsx, &tdata)) {
1528 pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, NULL, NULL);
1532 static int send_out_of_dialog_request(pjsip_tx_data *tdata, struct ast_sip_endpoint *endpoint)
1534 ao2_ref(endpoint, +1);
1535 if (pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, endpoint, send_request_cb) != PJ_SUCCESS) {
1536 ast_log(LOG_ERROR, "Error attempting to send outbound %.*s request to endpoint %s\n",
1537 (int) pj_strlen(&tdata->msg->line.req.method.name),
1538 pj_strbuf(&tdata->msg->line.req.method.name),
1539 ast_sorcery_object_get_id(endpoint));
1540 ao2_ref(endpoint, -1);
1547 int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint)
1549 ast_assert(tdata->msg->type == PJSIP_REQUEST_MSG);
1552 return send_in_dialog_request(tdata, dlg);
1554 return send_out_of_dialog_request(tdata, endpoint);
1558 int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value)
1562 pjsip_generic_string_hdr *hdr;
1564 pj_cstr(&hdr_name, name);
1565 pj_cstr(&hdr_value, value);
1567 hdr = pjsip_generic_string_hdr_create(tdata->pool, &hdr_name, &hdr_value);
1569 pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) hdr);
1573 static pjsip_msg_body *ast_body_to_pjsip_body(pj_pool_t *pool, const struct ast_sip_body *body)
1579 pj_cstr(&type, body->type);
1580 pj_cstr(&subtype, body->subtype);
1581 pj_cstr(&body_text, body->body_text);
1583 return pjsip_msg_body_create(pool, &type, &subtype, &body_text);
1586 int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body)
1588 pjsip_msg_body *pjsip_body = ast_body_to_pjsip_body(tdata->pool, body);
1589 tdata->msg->body = pjsip_body;
1593 int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies)
1596 /* NULL for type and subtype automatically creates "multipart/mixed" */
1597 pjsip_msg_body *body = pjsip_multipart_create(tdata->pool, NULL, NULL);
1599 for (i = 0; i < num_bodies; ++i) {
1600 pjsip_multipart_part *part = pjsip_multipart_create_part(tdata->pool);
1601 part->body = ast_body_to_pjsip_body(tdata->pool, bodies[i]);
1602 pjsip_multipart_add_part(tdata->pool, body, part);
1605 tdata->msg->body = body;
1609 int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text)
1611 size_t combined_size = strlen(body_text) + tdata->msg->body->len;
1612 struct ast_str *body_buffer = ast_str_alloca(combined_size);
1614 ast_str_set(&body_buffer, 0, "%.*s%s", (int) tdata->msg->body->len, (char *) tdata->msg->body->data, body_text);
1616 tdata->msg->body->data = pj_pool_alloc(tdata->pool, combined_size);
1617 pj_memcpy(tdata->msg->body->data, ast_str_buffer(body_buffer), combined_size);
1618 tdata->msg->body->len = combined_size;
1623 struct ast_taskprocessor *ast_sip_create_serializer(void)
1625 struct ast_taskprocessor *serializer;
1626 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1627 char name[AST_UUID_STR_LEN];
1633 ast_uuid_to_str(uuid, name, sizeof(name));
1635 serializer = ast_threadpool_serializer(name, sip_threadpool);
1642 int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1645 return ast_taskprocessor_push(serializer, sip_task, task_data);
1647 return ast_threadpool_push(sip_threadpool, sip_task, task_data);
1651 struct sync_task_data {
1656 int (*task)(void *);
1660 static int sync_task(void *data)
1662 struct sync_task_data *std = data;
1663 std->fail = std->task(std->task_data);
1665 ast_mutex_lock(&std->lock);
1667 ast_cond_signal(&std->cond);
1668 ast_mutex_unlock(&std->lock);
1672 int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1674 /* This method is an onion */
1675 struct sync_task_data std;
1676 ast_mutex_init(&std.lock);
1677 ast_cond_init(&std.cond, NULL);
1678 std.fail = std.complete = 0;
1679 std.task = sip_task;
1680 std.task_data = task_data;
1683 if (ast_taskprocessor_push(serializer, sync_task, &std)) {
1687 if (ast_threadpool_push(sip_threadpool, sync_task, &std)) {
1692 ast_mutex_lock(&std.lock);
1693 while (!std.complete) {
1694 ast_cond_wait(&std.cond, &std.lock);
1696 ast_mutex_unlock(&std.lock);
1698 ast_mutex_destroy(&std.lock);
1699 ast_cond_destroy(&std.cond);
1703 void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
1705 size_t chars_to_copy = MIN(size - 1, pj_strlen(src));
1706 memcpy(dest, pj_strbuf(src), chars_to_copy);
1707 dest[chars_to_copy] = '\0';
1710 int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype)
1712 pjsip_media_type compare;
1714 if (!content_type) {
1718 pjsip_media_type_init2(&compare, type, subtype);
1720 return pjsip_media_type_cmp(content_type, &compare, 0) ? 0 : -1;
1723 pj_caching_pool caching_pool;
1724 pj_pool_t *memory_pool;
1725 pj_thread_t *monitor_thread;
1726 static int monitor_continue;
1728 static void *monitor_thread_exec(void *endpt)
1730 while (monitor_continue) {
1731 const pj_time_val delay = {0, 10};
1732 pjsip_endpt_handle_events(ast_pjsip_endpoint, &delay);
1737 static void stop_monitor_thread(void)
1739 monitor_continue = 0;
1740 pj_thread_join(monitor_thread);
1743 AST_THREADSTORAGE(pj_thread_storage);
1744 AST_THREADSTORAGE(servant_id_storage);
1745 #define SIP_SERVANT_ID 0x5E2F1D
1747 static void sip_thread_start(void)
1749 pj_thread_desc *desc;
1750 pj_thread_t *thread;
1751 uint32_t *servant_id;
1753 servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1755 ast_log(LOG_ERROR, "Could not set SIP servant ID in thread-local storage.\n");
1758 *servant_id = SIP_SERVANT_ID;
1760 desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc));
1762 ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n");
1765 pj_bzero(*desc, sizeof(*desc));
1767 if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) {
1768 ast_log(LOG_ERROR, "Couldn't register thread with PJLIB.\n");
1772 int ast_sip_thread_is_servant(void)
1774 uint32_t *servant_id;
1776 servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1781 return *servant_id == SIP_SERVANT_ID;
1784 static void remove_request_headers(pjsip_endpoint *endpt)
1786 const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt);
1787 pjsip_hdr *iter = request_headers->next;
1789 while (iter != request_headers) {
1790 pjsip_hdr *to_erase = iter;
1792 pj_list_erase(to_erase);
1796 static int load_module(void)
1798 /* The third parameter is just copied from
1799 * example code from PJLIB. This can be adjusted
1803 struct ast_threadpool_options options;
1805 if (pj_init() != PJ_SUCCESS) {
1806 return AST_MODULE_LOAD_DECLINE;
1809 if (pjlib_util_init() != PJ_SUCCESS) {
1811 return AST_MODULE_LOAD_DECLINE;
1814 pj_caching_pool_init(&caching_pool, NULL, 1024 * 1024);
1815 if (pjsip_endpt_create(&caching_pool.factory, "SIP", &ast_pjsip_endpoint) != PJ_SUCCESS) {
1816 ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n");
1820 /* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that,
1821 * we need to stop PJSIP from doing it automatically
1823 remove_request_headers(ast_pjsip_endpoint);
1825 memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL);
1827 ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n");
1831 if (ast_sip_initialize_system()) {
1832 ast_log(LOG_ERROR, "Failed to initialize SIP system configuration. Aborting load\n");
1836 sip_get_threadpool_options(&options);
1837 options.thread_start = sip_thread_start;
1838 sip_threadpool = ast_threadpool_create("SIP", NULL, &options);
1839 if (!sip_threadpool) {
1840 ast_log(LOG_ERROR, "Failed to create SIP threadpool. Aborting load\n");
1844 pjsip_tsx_layer_init_module(ast_pjsip_endpoint);
1845 pjsip_ua_init_module(ast_pjsip_endpoint, NULL);
1847 monitor_continue = 1;
1848 status = pj_thread_create(memory_pool, "SIP", (pj_thread_proc *) &monitor_thread_exec,
1849 NULL, PJ_THREAD_DEFAULT_STACK_SIZE * 2, 0, &monitor_thread);
1850 if (status != PJ_SUCCESS) {
1851 ast_log(LOG_ERROR, "Failed to start SIP monitor thread. Aborting load\n");
1855 ast_sip_initialize_global_headers();
1857 if (ast_res_pjsip_initialize_configuration()) {
1858 ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
1862 if (ast_sip_initialize_distributor()) {
1863 ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n");
1867 if (ast_sip_initialize_outbound_authentication()) {
1868 ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n");
1872 ast_res_pjsip_init_options_handling(0);
1874 ast_res_pjsip_init_contact_transports();
1876 return AST_MODULE_LOAD_SUCCESS;
1879 ast_sip_destroy_distributor();
1880 ast_res_pjsip_destroy_configuration();
1881 ast_sip_destroy_global_headers();
1882 if (monitor_thread) {
1883 stop_monitor_thread();
1886 pj_pool_release(memory_pool);
1889 if (ast_pjsip_endpoint) {
1890 pjsip_endpt_destroy(ast_pjsip_endpoint);
1891 ast_pjsip_endpoint = NULL;
1893 pj_caching_pool_destroy(&caching_pool);
1894 return AST_MODULE_LOAD_DECLINE;
1897 static int reload_module(void)
1899 if (ast_res_pjsip_reload_configuration()) {
1900 return AST_MODULE_LOAD_DECLINE;
1902 ast_res_pjsip_init_options_handling(1);
1906 static int unload_pjsip(void *data)
1909 pj_pool_release(memory_pool);
1912 if (ast_pjsip_endpoint) {
1913 pjsip_endpt_destroy(ast_pjsip_endpoint);
1914 ast_pjsip_endpoint = NULL;
1916 pj_caching_pool_destroy(&caching_pool);
1920 static int unload_module(void)
1922 ast_res_pjsip_cleanup_options_handling();
1923 ast_sip_destroy_distributor();
1924 ast_res_pjsip_destroy_configuration();
1925 ast_sip_destroy_global_headers();
1926 if (monitor_thread) {
1927 stop_monitor_thread();
1929 /* The thread this is called from cannot call PJSIP/PJLIB functions,
1930 * so we have to push the work to the threadpool to handle
1932 ast_sip_push_task_synchronous(NULL, unload_pjsip, NULL);
1934 ast_threadpool_shutdown(sip_threadpool);
1939 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Basic SIP resource",
1940 .load = load_module,
1941 .unload = unload_module,
1942 .reload = reload_module,
1943 .load_pri = AST_MODPRI_CHANNEL_DEPEND - 5,
projects / asterisk/asterisk.git / blob