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.
217 Be aware that the <literal>external_media_address</literal> option, set in Transport
218 configuration, can also affect the final media address used in the SDP.
219 </note></description>
221 <configOption name="force_rport" default="yes">
222 <synopsis>Force use of return port</synopsis>
224 <configOption name="ice_support" default="no">
225 <synopsis>Enable the ICE mechanism to help traverse NAT</synopsis>
227 <configOption name="identify_by" default="username,location">
228 <synopsis>Way(s) for Endpoint to be identified</synopsis>
230 An endpoint can be identified in multiple ways. Currently, the only supported
231 option is <literal>username</literal>, which matches the endpoint based on the
232 username in the From header.
234 <note><para>Endpoints can also be identified by IP address; however, that method
235 of identification is not handled by this configuration option. See the documentation
236 for the <literal>identify</literal> configuration section for more details on that
237 method of endpoint identification. If this option is set to <literal>username</literal>
238 and an <literal>identify</literal> configuration section exists for the endpoint, then
239 the endpoint can be identified in multiple ways.</para></note>
241 <enum name="username" />
245 <configOption name="mailboxes">
246 <synopsis>Mailbox(es) to be associated with</synopsis>
248 <configOption name="mohsuggest" default="default">
249 <synopsis>Default Music On Hold class</synopsis>
251 <configOption name="outbound_auth">
252 <synopsis>Authentication object used for outbound requests</synopsis>
254 <configOption name="outbound_proxy">
255 <synopsis>Proxy through which to send requests</synopsis>
257 <configOption name="rewrite_contact">
258 <synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
260 On inbound SIP messages from this endpoint, the Contact header will be changed to have the
261 source IP address and port. This option does not affect outbound messages send to this
263 </para></description>
265 <configOption name="rtp_ipv6" default="no">
266 <synopsis>Allow use of IPv6 for RTP traffic</synopsis>
268 <configOption name="rtp_symmetric" default="no">
269 <synopsis>Enforce that RTP must be symmetric</synopsis>
271 <configOption name="send_diversion" default="yes">
272 <synopsis>Send the Diversion header, conveying the diversion
273 information to the called user agent</synopsis>
275 <configOption name="send_pai" default="no">
276 <synopsis>Send the P-Asserted-Identity header</synopsis>
278 <configOption name="send_rpid" default="no">
279 <synopsis>Send the Remote-Party-ID header</synopsis>
281 <configOption name="timers_min_se" default="90">
282 <synopsis>Minimum session timers expiration period</synopsis>
284 Minimium session timer expiration period. Time in seconds.
285 </para></description>
287 <configOption name="timers" default="yes">
288 <synopsis>Session timers for SIP packets</synopsis>
291 <enum name="forced" />
293 <enum name="required" />
298 <configOption name="timers_sess_expires" default="1800">
299 <synopsis>Maximum session timer expiration period</synopsis>
301 Maximium session timer expiration period. Time in seconds.
302 </para></description>
304 <configOption name="transport">
305 <synopsis>Desired transport configuration</synopsis>
307 This will set the desired transport configuration to send SIP data through.
309 <warning><para>Not specifying a transport will <emphasis>DEFAULT</emphasis>
310 to the first configured transport in <filename>pjsip.conf</filename> which is
311 valid for the URI we are trying to contact.
313 <warning><para>Transport configuration is not affected by reloads. In order to
314 change transports, a full Asterisk restart is required</para></warning>
317 <configOption name="trust_id_inbound" default="no">
318 <synopsis>Accept identification information received from this endpoint</synopsis>
319 <description><para>This option determines whether Asterisk will accept
320 identification from the endpoint from headers such as P-Asserted-Identity
321 or Remote-Party-ID header. This option applies both to calls originating from the
322 endpoint and calls originating from Asterisk. If <literal>no</literal>, the
323 configured Caller-ID from pjsip.conf will always be used as the identity for
324 the endpoint.</para></description>
326 <configOption name="trust_id_outbound" default="no">
327 <synopsis>Send private identification details to the endpoint.</synopsis>
328 <description><para>This option determines whether res_pjsip will send private
329 identification information to the endpoint. If <literal>no</literal>,
330 private Caller-ID information will not be forwarded to the endpoint.
331 "Private" in this case refers to any method of restricting identification.
332 Example: setting <replaceable>callerid_privacy</replaceable> to any
333 <literal>prohib</literal> variation.
334 Example: If <replaceable>trust_id_inbound</replaceable> is set to
335 <literal>yes</literal>, the presence of a <literal>Privacy: id</literal>
336 header in a SIP request or response would indicate the identification
337 provided in the request is private.</para></description>
339 <configOption name="type">
340 <synopsis>Must be of type 'endpoint'.</synopsis>
342 <configOption name="use_ptime" default="no">
343 <synopsis>Use Endpoint's requested packetisation interval</synopsis>
345 <configOption name="use_avpf" default="no">
346 <synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this
349 If set to <literal>yes</literal>, res_pjsip will use use the AVPF or SAVPF RTP
350 profile for all media offers on outbound calls and media updates and will
351 decline media offers not using the AVPF or SAVPF profile.
353 If set to <literal>no</literal>, res_pjsip will use use the AVP or SAVP RTP
354 profile for all media offers on outbound calls and media updates and will
355 decline media offers not using the AVP or SAVP profile.
356 </para></description>
358 <configOption name="media_encryption" default="no">
359 <synopsis>Determines whether res_pjsip will use and enforce usage of media encryption
360 for this endpoint.</synopsis>
363 <enum name="no"><para>
364 res_pjsip will offer no encryption and allow no encryption to be setup.
366 <enum name="sdes"><para>
367 res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP
368 transport should be used in conjunction with this option to prevent
369 exposure of media encryption keys.
371 <enum name="dtls"><para>
372 res_pjsip will offer DTLS-SRTP setup.
377 <configOption name="inband_progress" default="no">
378 <synopsis>Determines whether chan_pjsip will indicate ringing using inband
381 If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
382 when told to indicate ringing and will immediately start sending ringing
385 If set to <literal>no</literal>, chan_pjsip will send a 180 Ringing when told
386 to indicate ringing and will NOT send it as audio.
387 </para></description>
389 <configOption name="callgroup">
390 <synopsis>The numeric pickup groups for a channel.</synopsis>
392 Can be set to a comma separated list of numbers or ranges between the values
393 of 0-63 (maximum of 64 groups).
394 </para></description>
396 <configOption name="pickupgroup">
397 <synopsis>The numeric pickup groups that a channel can pickup.</synopsis>
399 Can be set to a comma separated list of numbers or ranges between the values
400 of 0-63 (maximum of 64 groups).
401 </para></description>
403 <configOption name="namedcallgroup">
404 <synopsis>The named pickup groups for a channel.</synopsis>
406 Can be set to a comma separated list of case sensitive strings limited by
407 supported line length.
408 </para></description>
410 <configOption name="namedpickupgroup">
411 <synopsis>The named pickup groups that a channel can pickup.</synopsis>
413 Can be set to a comma separated list of case sensitive strings limited by
414 supported line length.
415 </para></description>
417 <configOption name="devicestate_busy_at" default="0">
418 <synopsis>The number of in-use channels which will cause busy to be returned as device state</synopsis>
420 When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the
421 PJSIP channel driver will return busy as the device state instead of in use.
422 </para></description>
424 <configOption name="t38udptl" default="no">
425 <synopsis>Whether T.38 UDPTL support is enabled or not</synopsis>
427 If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted
429 </para></description>
431 <configOption name="t38udptl_ec" default="none">
432 <synopsis>T.38 UDPTL error correction method</synopsis>
435 <enum name="none"><para>
436 No error correction should be used.
438 <enum name="fec"><para>
439 Forward error correction should be used.
441 <enum name="redundancy"><para>
442 Redundacy error correction should be used.
447 <configOption name="t38udptl_maxdatagram" default="0">
448 <synopsis>T.38 UDPTL maximum datagram size</synopsis>
450 This option can be set to override the maximum datagram of a remote endpoint for broken
452 </para></description>
454 <configOption name="faxdetect" default="no">
455 <synopsis>Whether CNG tone detection is enabled</synopsis>
457 This option can be set to send the session to the fax extension when a CNG tone is
459 </para></description>
461 <configOption name="t38udptl_nat" default="no">
462 <synopsis>Whether NAT support is enabled on UDPTL sessions</synopsis>
464 When enabled the UDPTL stack will send UDPTL packets to the source address of
466 </para></description>
468 <configOption name="t38udptl_ipv6" default="no">
469 <synopsis>Whether IPv6 is used for UDPTL Sessions</synopsis>
471 When enabled the UDPTL stack will use IPv6.
472 </para></description>
474 <configOption name="tonezone">
475 <synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis>
477 <configOption name="language">
478 <synopsis>Set the default language to use for channels created for this endpoint.</synopsis>
480 <configOption name="one_touch_recording" default="no">
481 <synopsis>Determines whether one-touch recording is allowed for this endpoint.</synopsis>
483 <ref type="configOption">recordonfeature</ref>
484 <ref type="configOption">recordofffeature</ref>
487 <configOption name="recordonfeature" default="automixmon">
488 <synopsis>The feature to enact when one-touch recording is turned on.</synopsis>
490 <para>When an INFO request for one-touch recording arrives with a Record header set to "on", this
491 feature will be enabled for the channel. The feature designated here can be any built-in
492 or dynamic feature defined in features.conf.</para>
493 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
496 <ref type="configOption">one_touch_recording</ref>
497 <ref type="configOption">recordofffeature</ref>
500 <configOption name="recordofffeature" default="automixmon">
501 <synopsis>The feature to enact when one-touch recording is turned off.</synopsis>
503 <para>When an INFO request for one-touch recording arrives with a Record header set to "off", this
504 feature will be enabled for the channel. The feature designated here can be any built-in
505 or dynamic feature defined in features.conf.</para>
506 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
509 <ref type="configOption">one_touch_recording</ref>
510 <ref type="configOption">recordonfeature</ref>
513 <configOption name="rtpengine" default="asterisk">
514 <synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis>
516 <configOption name="allowtransfer" default="yes">
517 <synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis>
519 <configOption name="sdpowner" default="-">
520 <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
522 <configOption name="sdpsession" default="Asterisk">
523 <synopsis>String used for the SDP session (s=) line.</synopsis>
525 <configOption name="tos_audio">
526 <synopsis>DSCP TOS bits 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="tos_video">
532 <synopsis>DSCP TOS bits 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="cos_audio">
538 <synopsis>Priority for audio streams</synopsis>
540 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
541 </para></description>
543 <configOption name="cos_video">
544 <synopsis>Priority for video streams</synopsis>
546 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
547 </para></description>
549 <configOption name="allowsubscribe" default="yes">
550 <synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis>
552 <configOption name="subminexpiry" default="60">
553 <synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis>
555 <configOption name="fromuser">
556 <synopsis>Username to use in From header for requests to this endpoint.</synopsis>
558 <configOption name="mwifromuser">
559 <synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis>
561 <configOption name="fromdomain">
562 <synopsis>Domain to user in From header for requests to this endpoint.</synopsis>
564 <configOption name="dtlsverify">
565 <synopsis>Verify that the provided peer certificate is valid</synopsis>
567 This option only applies if <replaceable>media_encryption</replaceable> is
568 set to <literal>dtls</literal>.
569 </para></description>
571 <configOption name="dtlsrekey">
572 <synopsis>Interval at which to renegotiate the TLS session and rekey the SRTP session</synopsis>
574 This option only applies if <replaceable>media_encryption</replaceable> is
575 set to <literal>dtls</literal>.
577 If this is not set or the value provided is 0 rekeying will be disabled.
578 </para></description>
580 <configOption name="dtlscertfile">
581 <synopsis>Path to certificate file to present to peer</synopsis>
583 This option only applies if <replaceable>media_encryption</replaceable> is
584 set to <literal>dtls</literal>.
585 </para></description>
587 <configOption name="dtlsprivatekey">
588 <synopsis>Path to private key for certificate file</synopsis>
590 This option only applies if <replaceable>media_encryption</replaceable> is
591 set to <literal>dtls</literal>.
592 </para></description>
594 <configOption name="dtlscipher">
595 <synopsis>Cipher to use for DTLS negotiation</synopsis>
597 This option only applies if <replaceable>media_encryption</replaceable> is
598 set to <literal>dtls</literal>.
600 Many options for acceptable ciphers. See link for more:
601 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
602 </para></description>
604 <configOption name="dtlscafile">
605 <synopsis>Path to certificate authority certificate</synopsis>
607 This option only applies if <replaceable>media_encryption</replaceable> is
608 set to <literal>dtls</literal>.
609 </para></description>
611 <configOption name="dtlscapath">
612 <synopsis>Path to a directory containing certificate authority certificates</synopsis>
614 This option only applies if <replaceable>media_encryption</replaceable> is
615 set to <literal>dtls</literal>.
616 </para></description>
618 <configOption name="dtlssetup">
619 <synopsis>Whether we are willing to accept connections, connect to the other party, or both.</synopsis>
622 This option only applies if <replaceable>media_encryption</replaceable> is
623 set to <literal>dtls</literal>.
626 <enum name="active"><para>
627 res_pjsip will make a connection to the peer.
629 <enum name="passive"><para>
630 res_pjsip will accept connections from the peer.
632 <enum name="actpass"><para>
633 res_pjsip will offer and accept connections from the peer.
638 <configOption name="srtp_tag_32">
639 <synopsis>Determines whether 32 byte tags should be used instead of 80 byte tags.</synopsis>
641 This option only applies if <replaceable>media_encryption</replaceable> is
642 set to <literal>sdes</literal> or <literal>dtls</literal>.
643 </para></description>
646 <configObject name="auth">
647 <synopsis>Authentication type</synopsis>
649 Authentication objects hold the authentication information for use
650 by other objects such as <literal>endpoints</literal> or <literal>registrations</literal>.
651 This also allows for multiple objects to use a single auth object. See
652 the <literal>auth_type</literal> config option for password style choices.
653 </para></description>
654 <configOption name="auth_type" default="userpass">
655 <synopsis>Authentication type</synopsis>
657 This option specifies which of the password style config options should be read
658 when trying to authenticate an endpoint inbound request. If set to <literal>userpass</literal>
659 then we'll read from the 'password' option. For <literal>md5</literal> we'll read
664 <enum name="userpass"/>
668 <configOption name="nonce_lifetime" default="32">
669 <synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
671 <configOption name="md5_cred">
672 <synopsis>MD5 Hash used for authentication.</synopsis>
673 <description><para>Only used when auth_type is <literal>md5</literal>.</para></description>
675 <configOption name="password">
676 <synopsis>PlainText password used for authentication.</synopsis>
677 <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
679 <configOption name="realm" default="asterisk">
680 <synopsis>SIP realm for endpoint</synopsis>
682 <configOption name="type">
683 <synopsis>Must be 'auth'</synopsis>
685 <configOption name="username">
686 <synopsis>Username to use for account</synopsis>
689 <configObject name="domain_alias">
690 <synopsis>Domain Alias</synopsis>
692 Signifies that a domain is an alias. If the domain on a session is
693 not found to match an AoR then this object is used to see if we have
694 an alias for the AoR to which the endpoint is binding. This objects
695 name as defined in configuration should be the domain alias and a
696 config option is provided to specify the domain to be aliased.
697 </para></description>
698 <configOption name="type">
699 <synopsis>Must be of type 'domain_alias'.</synopsis>
701 <configOption name="domain">
702 <synopsis>Domain to be aliased</synopsis>
705 <configObject name="transport">
706 <synopsis>SIP Transport</synopsis>
708 <emphasis>Transports</emphasis>
710 <para>There are different transports and protocol derivatives
711 supported by <literal>res_pjsip</literal>. They are in order of
712 preference: UDP, TCP, and WebSocket (WS).</para>
713 <note><para>Changes to transport configuration in pjsip.conf will only be
714 effected on a complete restart of Asterisk. A module reload
715 will not suffice.</para></note>
717 <configOption name="async_operations" default="1">
718 <synopsis>Number of simultaneous Asynchronous Operations</synopsis>
720 <configOption name="bind">
721 <synopsis>IP Address and optional port to bind to for this transport</synopsis>
723 <configOption name="ca_list_file">
724 <synopsis>File containing a list of certificates to read (TLS ONLY)</synopsis>
726 <configOption name="cert_file">
727 <synopsis>Certificate file for endpoint (TLS ONLY)</synopsis>
729 <configOption name="cipher">
730 <synopsis>Preferred Cryptography Cipher (TLS ONLY)</synopsis>
732 Many options for acceptable ciphers see link for more:
733 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
734 </para></description>
736 <configOption name="domain">
737 <synopsis>Domain the transport comes from</synopsis>
739 <configOption name="external_media_address">
740 <synopsis>External IP address to use in RTP handling</synopsis>
742 When a request or response is sent out, if the destination of the
743 message is outside the IP network defined in the option <literal>localnet</literal>,
744 and the media address in the SDP is within the localnet network, then the
745 media address in the SDP will be rewritten to the value defined for
746 <literal>external_media_address</literal>.
747 </para></description>
749 <configOption name="external_signaling_address">
750 <synopsis>External address for SIP signalling</synopsis>
752 <configOption name="external_signaling_port" default="0">
753 <synopsis>External port for SIP signalling</synopsis>
755 <configOption name="method">
756 <synopsis>Method of SSL transport (TLS ONLY)</synopsis>
759 <enum name="default" />
760 <enum name="unspecified" />
761 <enum name="tlsv1" />
762 <enum name="sslv2" />
763 <enum name="sslv3" />
764 <enum name="sslv23" />
768 <configOption name="localnet">
769 <synopsis>Network to consider local (used for NAT purposes).</synopsis>
770 <description><para>This must be in CIDR or dotted decimal format with the IP
771 and mask separated with a slash ('/').</para></description>
773 <configOption name="password">
774 <synopsis>Password required for transport</synopsis>
776 <configOption name="privkey_file">
777 <synopsis>Private key file (TLS ONLY)</synopsis>
779 <configOption name="protocol" default="udp">
780 <synopsis>Protocol to use for SIP traffic</synopsis>
791 <configOption name="require_client_cert" default="false">
792 <synopsis>Require client certificate (TLS ONLY)</synopsis>
794 <configOption name="type">
795 <synopsis>Must be of type 'transport'.</synopsis>
797 <configOption name="verify_client" default="false">
798 <synopsis>Require verification of client certificate (TLS ONLY)</synopsis>
800 <configOption name="verify_server" default="false">
801 <synopsis>Require verification of server certificate (TLS ONLY)</synopsis>
803 <configOption name="tos" default="false">
804 <synopsis>Enable TOS for the signalling sent over this transport</synopsis>
806 <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
807 for more information on this parameter.</para>
808 <note><para>This option does not apply to the <replaceable>ws</replaceable>
809 or the <replaceable>wss</replaceable> protocols.</para></note>
812 <configOption name="cos" default="false">
813 <synopsis>Enable COS for the signalling sent over this transport</synopsis>
815 <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
816 for more information on this parameter.</para>
817 <note><para>This option does not apply to the <replaceable>ws</replaceable>
818 or the <replaceable>wss</replaceable> protocols.</para></note>
822 <configObject name="contact">
823 <synopsis>A way of creating an aliased name to a SIP URI</synopsis>
825 Contacts are a way to hide SIP URIs from the dialplan directly.
826 They are also used to make a group of contactable parties when
827 in use with <literal>AoR</literal> lists.
828 </para></description>
829 <configOption name="type">
830 <synopsis>Must be of type 'contact'.</synopsis>
832 <configOption name="uri">
833 <synopsis>SIP URI to contact peer</synopsis>
835 <configOption name="expiration_time">
836 <synopsis>Time to keep alive a contact</synopsis>
838 Time to keep alive a contact. String style specification.
839 </para></description>
841 <configOption name="qualify_frequency" default="0">
842 <synopsis>Interval at which to qualify a contact</synopsis>
844 Interval between attempts to qualify the contact for reachability.
845 If <literal>0</literal> never qualify. Time in seconds.
846 </para></description>
849 <configObject name="aor">
850 <synopsis>The configuration for a location of an endpoint</synopsis>
852 An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
853 AoRs are specified, an endpoint will not be reachable by Asterisk.
854 Beyond that, an AoR has other uses within Asterisk, such as inbound
857 An <literal>AoR</literal> is a way to allow dialing a group
858 of <literal>Contacts</literal> that all use the same
859 <literal>endpoint</literal> for calls.
861 This can be used as another way of grouping a list of contacts to dial
862 rather than specifing them each directly when dialing via the dialplan.
863 This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
865 Registrations: For Asterisk to match an inbound registration to an endpoint,
866 the AoR object name must match the user portion of the SIP URI in the "To:"
867 header of the inbound SIP registration. That will usually be equivalent
868 to the "user name" set in your hard or soft phones configuration.
869 </para></description>
870 <configOption name="contact">
871 <synopsis>Permanent contacts assigned to AoR</synopsis>
873 Contacts specified will be called whenever referenced
874 by <literal>chan_pjsip</literal>.
876 Use a separate "contact=" entry for each contact required. Contacts
877 are specified using a SIP URI.
878 </para></description>
880 <configOption name="default_expiration" default="3600">
881 <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
883 <configOption name="mailboxes">
884 <synopsis>Mailbox(es) to be associated with</synopsis>
885 <description><para>This option applies when an external entity subscribes to an AoR
886 for message waiting indications. The mailboxes specified will be subscribed to.
887 More than one mailbox can be specified with a comma-delimited string.</para></description>
889 <configOption name="maximum_expiration" default="7200">
890 <synopsis>Maximum time to keep an AoR</synopsis>
892 Maximium time to keep a peer with explicit expiration. Time in seconds.
893 </para></description>
895 <configOption name="max_contacts" default="0">
896 <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
898 Maximum number of contacts that can associate with this AoR. This value does
899 not affect the number of contacts that can be added with the "contact" option.
900 It only limits contacts added through external interaction, such as
903 <note><para>This should be set to <literal>1</literal> and
904 <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
905 wish to stick with the older <literal>chan_sip</literal> behaviour.
909 <configOption name="minimum_expiration" default="60">
910 <synopsis>Minimum keep alive time for an AoR</synopsis>
912 Minimum time to keep a peer with an explict expiration. Time in seconds.
913 </para></description>
915 <configOption name="remove_existing" default="no">
916 <synopsis>Determines whether new contacts replace existing ones.</synopsis>
918 On receiving a new registration to the AoR should it remove
919 the existing contact that was registered against it?
921 <note><para>This should be set to <literal>yes</literal> and
922 <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
923 wish to stick with the older <literal>chan_sip</literal> behaviour.
927 <configOption name="type">
928 <synopsis>Must be of type 'aor'.</synopsis>
930 <configOption name="qualify_frequency" default="0">
931 <synopsis>Interval at which to qualify an AoR</synopsis>
933 Interval between attempts to qualify the AoR for reachability.
934 If <literal>0</literal> never qualify. Time in seconds.
935 </para></description>
937 <configOption name="authenticate_qualify" default="no">
938 <synopsis>Authenticates a qualify request if needed</synopsis>
940 If true and a qualify request receives a challenge or authenticate response
941 authentication is attempted before declaring the contact available.
942 </para></description>
945 <configObject name="system">
946 <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
948 The settings in this section are global. In addition to being global, the values will
949 not be re-evaluated when a reload is performed. This is because the values must be set
950 before the SIP stack is initialized. The only way to reset these values is to either
951 restart Asterisk, or unload res_pjsip.so and then load it again.
952 </para></description>
953 <configOption name="timert1" default="500">
954 <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
956 Timer T1 is the base for determining how long to wait before retransmitting
957 requests that receive no response when using an unreliable transport (e.g. UDP).
958 For more information on this timer, see RFC 3261, Section 17.1.1.1.
959 </para></description>
961 <configOption name="timerb" default="32000">
962 <synopsis>Set transaction timer B value (milliseconds).</synopsis>
964 Timer B determines the maximum amount of time to wait after sending an INVITE
965 request before terminating the transaction. It is recommended that this be set
966 to 64 * Timer T1, but it may be set higher if desired. For more information on
967 this timer, see RFC 3261, Section 17.1.1.1.
968 </para></description>
970 <configOption name="compactheaders" default="no">
971 <synopsis>Use the short forms of common SIP header names.</synopsis>
973 <configOption name="threadpool_initial_size" default="0">
974 <synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
976 <configOption name="threadpool_auto_increment" default="5">
977 <synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
979 <configOption name="threadpool_idle_timeout" default="60">
980 <synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
982 <configOption name="threadpool_max_size" default="0">
983 <synopsis>Maximum number of threads in the res_pjsip threadpool.
984 A value of 0 indicates no maximum.</synopsis>
986 <configOption name="type">
987 <synopsis>Must be of type 'system'.</synopsis>
990 <configObject name="global">
991 <synopsis>Options that apply globally to all SIP communications</synopsis>
993 The settings in this section are global. Unlike options in the <literal>system</literal>
994 section, these options can be refreshed by performing a reload.
995 </para></description>
996 <configOption name="maxforwards" default="70">
997 <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
999 <configOption name="type">
1000 <synopsis>Must be of type 'global'.</synopsis>
1002 <configOption name="useragent" default="Asterisk <Asterisk Version>">
1003 <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
1008 <manager name="PJSIPQualify" language="en_US">
1010 Qualify a chan_pjsip endpoint.
1013 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1014 <parameter name="Endpoint" required="true">
1015 <para>The endpoint you want to qualify.</para>
1019 <para>Qualify a chan_pjsip endpoint.</para>
1025 static pjsip_endpoint *ast_pjsip_endpoint;
1027 static struct ast_threadpool *sip_threadpool;
1029 static int register_service(void *data)
1031 pjsip_module **module = data;
1032 if (!ast_pjsip_endpoint) {
1033 ast_log(LOG_ERROR, "There is no PJSIP endpoint. Unable to register services\n");
1036 if (pjsip_endpt_register_module(ast_pjsip_endpoint, *module) != PJ_SUCCESS) {
1037 ast_log(LOG_ERROR, "Unable to register module %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1040 ast_debug(1, "Registered SIP service %.*s (%p)\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name), *module);
1041 ast_module_ref(ast_module_info->self);
1045 int ast_sip_register_service(pjsip_module *module)
1047 return ast_sip_push_task_synchronous(NULL, register_service, &module);
1050 static int unregister_service(void *data)
1052 pjsip_module **module = data;
1053 ast_module_unref(ast_module_info->self);
1054 if (!ast_pjsip_endpoint) {
1057 pjsip_endpt_unregister_module(ast_pjsip_endpoint, *module);
1058 ast_debug(1, "Unregistered SIP service %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1062 void ast_sip_unregister_service(pjsip_module *module)
1064 ast_sip_push_task_synchronous(NULL, unregister_service, &module);
1067 static struct ast_sip_authenticator *registered_authenticator;
1069 int ast_sip_register_authenticator(struct ast_sip_authenticator *auth)
1071 if (registered_authenticator) {
1072 ast_log(LOG_WARNING, "Authenticator %p is already registered. Cannot register a new one\n", registered_authenticator);
1075 registered_authenticator = auth;
1076 ast_debug(1, "Registered SIP authenticator module %p\n", auth);
1077 ast_module_ref(ast_module_info->self);
1081 void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth)
1083 if (registered_authenticator != auth) {
1084 ast_log(LOG_WARNING, "Trying to unregister authenticator %p but authenticator %p registered\n",
1085 auth, registered_authenticator);
1088 registered_authenticator = NULL;
1089 ast_debug(1, "Unregistered SIP authenticator %p\n", auth);
1090 ast_module_unref(ast_module_info->self);
1093 int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1095 if (!registered_authenticator) {
1096 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is not required\n");
1100 return registered_authenticator->requires_authentication(endpoint, rdata);
1103 enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
1104 pjsip_rx_data *rdata, pjsip_tx_data *tdata)
1106 if (!registered_authenticator) {
1107 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is successful\n");
1110 return registered_authenticator->check_authentication(endpoint, rdata, tdata);
1113 static struct ast_sip_outbound_authenticator *registered_outbound_authenticator;
1115 int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1117 if (registered_outbound_authenticator) {
1118 ast_log(LOG_WARNING, "Outbound authenticator %p is already registered. Cannot register a new one\n", registered_outbound_authenticator);
1121 registered_outbound_authenticator = auth;
1122 ast_debug(1, "Registered SIP outbound authenticator module %p\n", auth);
1123 ast_module_ref(ast_module_info->self);
1127 void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1129 if (registered_outbound_authenticator != auth) {
1130 ast_log(LOG_WARNING, "Trying to unregister outbound authenticator %p but outbound authenticator %p registered\n",
1131 auth, registered_outbound_authenticator);
1134 registered_outbound_authenticator = NULL;
1135 ast_debug(1, "Unregistered SIP outbound authenticator %p\n", auth);
1136 ast_module_unref(ast_module_info->self);
1139 int ast_sip_create_request_with_auth(const struct ast_sip_auth_array *auths, pjsip_rx_data *challenge,
1140 pjsip_transaction *tsx, pjsip_tx_data **new_request)
1142 if (!registered_outbound_authenticator) {
1143 ast_log(LOG_WARNING, "No SIP outbound authenticator registered. Cannot respond to authentication challenge\n");
1146 return registered_outbound_authenticator->create_request_with_auth(auths, challenge, tsx, new_request);
1149 struct endpoint_identifier_list {
1150 struct ast_sip_endpoint_identifier *identifier;
1151 AST_RWLIST_ENTRY(endpoint_identifier_list) list;
1154 static AST_RWLIST_HEAD_STATIC(endpoint_identifiers, endpoint_identifier_list);
1156 int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1158 struct endpoint_identifier_list *id_list_item;
1159 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1161 id_list_item = ast_calloc(1, sizeof(*id_list_item));
1162 if (!id_list_item) {
1163 ast_log(LOG_ERROR, "Unabled to add endpoint identifier. Out of memory.\n");
1166 id_list_item->identifier = identifier;
1168 AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list);
1169 ast_debug(1, "Registered endpoint identifier %p\n", identifier);
1171 ast_module_ref(ast_module_info->self);
1175 void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1177 struct endpoint_identifier_list *iter;
1178 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1179 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) {
1180 if (iter->identifier == identifier) {
1181 AST_RWLIST_REMOVE_CURRENT(list);
1183 ast_debug(1, "Unregistered endpoint identifier %p\n", identifier);
1184 ast_module_unref(ast_module_info->self);
1188 AST_RWLIST_TRAVERSE_SAFE_END;
1191 struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata)
1193 struct endpoint_identifier_list *iter;
1194 struct ast_sip_endpoint *endpoint = NULL;
1195 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1196 AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) {
1197 ast_assert(iter->identifier->identify_endpoint != NULL);
1198 endpoint = iter->identifier->identify_endpoint(rdata);
1206 pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
1208 return ast_pjsip_endpoint;
1211 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)
1213 pj_str_t tmp, local_addr;
1215 pjsip_sip_uri *sip_uri;
1216 pjsip_transport_type_e type = PJSIP_TRANSPORT_UNSPECIFIED;
1218 char uuid_str[AST_UUID_STR_LEN];
1220 if (ast_strlen_zero(user)) {
1221 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1225 user = ast_uuid_to_str(uuid, uuid_str, sizeof(uuid_str));
1228 /* Parse the provided target URI so we can determine what transport it will end up using */
1229 pj_strdup_with_null(pool, &tmp, target);
1231 if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) ||
1232 (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
1236 sip_uri = pjsip_uri_get_uri(uri);
1238 /* Determine the transport type to use */
1239 if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
1240 type = PJSIP_TRANSPORT_TLS;
1241 } else if (!sip_uri->transport_param.slen) {
1242 type = PJSIP_TRANSPORT_UDP;
1244 type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
1247 if (type == PJSIP_TRANSPORT_UNSPECIFIED) {
1251 /* If the host is IPv6 turn the transport into an IPv6 version */
1252 if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1253 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1256 if (!ast_strlen_zero(domain)) {
1257 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1258 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1260 (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1263 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1264 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1268 /* Get the local bound address for the transport that will be used when communicating with the provided URI */
1269 if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), pool, type, selector,
1270 &local_addr, &local_port) != PJ_SUCCESS) {
1274 /* If IPv6 was specified in the transport, set the proper type */
1275 if (pj_strchr(&local_addr, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1276 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1279 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1280 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1281 "<%s:%s@%s%.*s%s:%d%s%s>",
1282 (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1284 (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1285 (int)local_addr.slen,
1287 (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1289 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1290 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1295 static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
1297 RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
1298 const char *transport_name = endpoint->transport;
1300 if (ast_strlen_zero(transport_name)) {
1304 transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name);
1306 if (!transport || !transport->state) {
1307 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport '%s' for endpoint '%s'\n",
1308 transport_name, ast_sorcery_object_get_id(endpoint));
1312 if (transport->state->transport) {
1313 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1314 selector->u.transport = transport->state->transport;
1315 } else if (transport->state->factory) {
1316 selector->type = PJSIP_TPSELECTOR_LISTENER;
1317 selector->u.listener = transport->state->factory;
1325 static int sip_get_tpselector_from_uri(const char *uri, pjsip_tpselector *selector)
1327 RAII_VAR(struct ast_sip_contact_transport *, contact_transport, NULL, ao2_cleanup);
1329 contact_transport = ast_sip_location_retrieve_contact_transport_by_uri(uri);
1331 if (!contact_transport) {
1335 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1336 selector->u.transport = contact_transport->transport;
1341 pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
1343 pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri;
1344 pjsip_dialog *dlg = NULL;
1345 const char *outbound_proxy = endpoint->outbound_proxy;
1346 pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1347 static const pj_str_t HCONTACT = { "Contact", 7 };
1349 pj_cstr(&remote_uri, uri);
1351 if (pjsip_dlg_create_uac(pjsip_ua_instance(), &local_uri, NULL, &remote_uri, NULL, &dlg) != PJ_SUCCESS) {
1355 if (sip_get_tpselector_from_uri(uri, &selector) && sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1356 pjsip_dlg_terminate(dlg);
1360 if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
1361 pjsip_dlg_terminate(dlg);
1365 /* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
1366 pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
1367 dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
1368 dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
1370 /* If a request user has been specified and we are permitted to change it, do so */
1371 if (!ast_strlen_zero(request_user) && (PJSIP_URI_SCHEME_IS_SIP(dlg->target) || PJSIP_URI_SCHEME_IS_SIPS(dlg->target))) {
1372 pjsip_sip_uri *target = pjsip_uri_get_uri(dlg->target);
1373 pj_strdup2(dlg->pool, &target->user, request_user);
1376 /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
1379 pjsip_dlg_set_transport(dlg, &selector);
1381 if (!ast_strlen_zero(outbound_proxy)) {
1382 pjsip_route_hdr route_set, *route;
1383 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1386 pj_list_init(&route_set);
1388 pj_strdup2_with_null(dlg->pool, &tmp, outbound_proxy);
1389 if (!(route = pjsip_parse_hdr(dlg->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1390 pjsip_dlg_terminate(dlg);
1393 pj_list_push_back(&route_set, route);
1395 pjsip_dlg_set_route_set(dlg, &route_set);
1403 pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1407 pjsip_transport_type_e type = rdata->tp_info.transport->key.type;
1410 contact.ptr = pj_pool_alloc(rdata->tp_info.pool, PJSIP_MAX_URL_SIZE);
1411 contact.slen = pj_ansi_snprintf(contact.ptr, PJSIP_MAX_URL_SIZE,
1412 "<%s:%s%.*s%s:%d%s%s>",
1413 (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1414 (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1415 (int)rdata->tp_info.transport->local_name.host.slen,
1416 rdata->tp_info.transport->local_name.host.ptr,
1417 (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1418 rdata->tp_info.transport->local_name.port,
1419 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1420 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1422 status = pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, &contact, &dlg);
1423 if (status != PJ_SUCCESS) {
1424 char err[PJ_ERR_MSG_SIZE];
1426 pjsip_strerror(status, err, sizeof(err));
1427 ast_log(LOG_ERROR, "Could not create dialog with endpoint %s. %s\n",
1428 ast_sorcery_object_get_id(endpoint), err);
1435 /* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
1436 static const pjsip_method info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
1437 static const pjsip_method message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
1441 const pjsip_method *pmethod;
1443 { "INVITE", &pjsip_invite_method },
1444 { "CANCEL", &pjsip_cancel_method },
1445 { "ACK", &pjsip_ack_method },
1446 { "BYE", &pjsip_bye_method },
1447 { "REGISTER", &pjsip_register_method },
1448 { "OPTIONS", &pjsip_options_method },
1449 { "SUBSCRIBE", &pjsip_subscribe_method },
1450 { "NOTIFY", &pjsip_notify_method },
1451 { "PUBLISH", &pjsip_publish_method },
1452 { "INFO", &info_method },
1453 { "MESSAGE", &message_method },
1456 static const pjsip_method *get_pjsip_method(const char *method)
1459 for (i = 0; i < ARRAY_LEN(methods); ++i) {
1460 if (!strcmp(method, methods[i].method)) {
1461 return methods[i].pmethod;
1467 static int create_in_dialog_request(const pjsip_method *method, struct pjsip_dialog *dlg, pjsip_tx_data **tdata)
1469 if (pjsip_dlg_create_request(dlg, method, -1, tdata) != PJ_SUCCESS) {
1470 ast_log(LOG_WARNING, "Unable to create in-dialog request.\n");
1477 static int create_out_of_dialog_request(const pjsip_method *method, struct ast_sip_endpoint *endpoint,
1478 const char *uri, pjsip_tx_data **tdata)
1480 RAII_VAR(struct ast_sip_contact *, contact, NULL, ao2_cleanup);
1481 pj_str_t remote_uri;
1484 pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1486 if (ast_strlen_zero(uri)) {
1488 ast_log(LOG_ERROR, "An endpoint and/or uri must be specified\n");
1492 contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
1493 if (!contact || ast_strlen_zero(contact->uri)) {
1494 ast_log(LOG_ERROR, "Unable to retrieve contact for endpoint %s\n",
1495 ast_sorcery_object_get_id(endpoint));
1499 pj_cstr(&remote_uri, contact->uri);
1501 pj_cstr(&remote_uri, uri);
1505 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1506 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
1507 ast_sorcery_object_get_id(endpoint));
1512 pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
1515 ast_log(LOG_ERROR, "Unable to create PJLIB memory pool\n");
1519 if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL,
1520 endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
1521 ast_log(LOG_ERROR, "Unable to create From header for %.*s request to endpoint %s\n",
1522 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1523 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1527 if (pjsip_endpt_create_request(ast_sip_get_pjsip_endpoint(), method, &remote_uri,
1528 &from, &remote_uri, &from, NULL, -1, NULL, tdata) != PJ_SUCCESS) {
1529 ast_log(LOG_ERROR, "Unable to create outbound %.*s request to endpoint %s\n",
1530 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1531 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1535 /* We can release this pool since request creation copied all the necessary
1536 * data into the outbound request's pool
1538 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1542 int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
1543 struct ast_sip_endpoint *endpoint, const char *uri,
1544 pjsip_tx_data **tdata)
1546 const pjsip_method *pmethod = get_pjsip_method(method);
1549 ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
1554 return create_in_dialog_request(pmethod, dlg, tdata);
1556 return create_out_of_dialog_request(pmethod, endpoint, uri, tdata);
1560 static int send_in_dialog_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg)
1562 if (pjsip_dlg_send_request(dlg, tdata, -1, NULL) != PJ_SUCCESS) {
1563 ast_log(LOG_WARNING, "Unable to send in-dialog request.\n");
1569 static void send_request_cb(void *token, pjsip_event *e)
1571 RAII_VAR(struct ast_sip_endpoint *, endpoint, token, ao2_cleanup);
1572 pjsip_transaction *tsx = e->body.tsx_state.tsx;
1573 pjsip_rx_data *challenge = e->body.tsx_state.src.rdata;
1574 pjsip_tx_data *tdata;
1576 if (tsx->status_code != 401 && tsx->status_code != 407) {
1580 if (!ast_sip_create_request_with_auth(&endpoint->outbound_auths, challenge, tsx, &tdata)) {
1581 pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, NULL, NULL);
1585 static int send_out_of_dialog_request(pjsip_tx_data *tdata, struct ast_sip_endpoint *endpoint)
1587 ao2_ref(endpoint, +1);
1588 if (pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, endpoint, send_request_cb) != PJ_SUCCESS) {
1589 ast_log(LOG_ERROR, "Error attempting to send outbound %.*s request to endpoint %s\n",
1590 (int) pj_strlen(&tdata->msg->line.req.method.name),
1591 pj_strbuf(&tdata->msg->line.req.method.name),
1592 ast_sorcery_object_get_id(endpoint));
1593 ao2_ref(endpoint, -1);
1600 int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint)
1602 ast_assert(tdata->msg->type == PJSIP_REQUEST_MSG);
1605 return send_in_dialog_request(tdata, dlg);
1607 return send_out_of_dialog_request(tdata, endpoint);
1611 int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value)
1615 pjsip_generic_string_hdr *hdr;
1617 pj_cstr(&hdr_name, name);
1618 pj_cstr(&hdr_value, value);
1620 hdr = pjsip_generic_string_hdr_create(tdata->pool, &hdr_name, &hdr_value);
1622 pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) hdr);
1626 static pjsip_msg_body *ast_body_to_pjsip_body(pj_pool_t *pool, const struct ast_sip_body *body)
1632 pj_cstr(&type, body->type);
1633 pj_cstr(&subtype, body->subtype);
1634 pj_cstr(&body_text, body->body_text);
1636 return pjsip_msg_body_create(pool, &type, &subtype, &body_text);
1639 int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body)
1641 pjsip_msg_body *pjsip_body = ast_body_to_pjsip_body(tdata->pool, body);
1642 tdata->msg->body = pjsip_body;
1646 int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies)
1649 /* NULL for type and subtype automatically creates "multipart/mixed" */
1650 pjsip_msg_body *body = pjsip_multipart_create(tdata->pool, NULL, NULL);
1652 for (i = 0; i < num_bodies; ++i) {
1653 pjsip_multipart_part *part = pjsip_multipart_create_part(tdata->pool);
1654 part->body = ast_body_to_pjsip_body(tdata->pool, bodies[i]);
1655 pjsip_multipart_add_part(tdata->pool, body, part);
1658 tdata->msg->body = body;
1662 int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text)
1664 size_t combined_size = strlen(body_text) + tdata->msg->body->len;
1665 struct ast_str *body_buffer = ast_str_alloca(combined_size);
1667 ast_str_set(&body_buffer, 0, "%.*s%s", (int) tdata->msg->body->len, (char *) tdata->msg->body->data, body_text);
1669 tdata->msg->body->data = pj_pool_alloc(tdata->pool, combined_size);
1670 pj_memcpy(tdata->msg->body->data, ast_str_buffer(body_buffer), combined_size);
1671 tdata->msg->body->len = combined_size;
1676 struct ast_taskprocessor *ast_sip_create_serializer(void)
1678 struct ast_taskprocessor *serializer;
1679 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1680 char name[AST_UUID_STR_LEN];
1686 ast_uuid_to_str(uuid, name, sizeof(name));
1688 serializer = ast_threadpool_serializer(name, sip_threadpool);
1695 int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1698 return ast_taskprocessor_push(serializer, sip_task, task_data);
1700 return ast_threadpool_push(sip_threadpool, sip_task, task_data);
1704 struct sync_task_data {
1709 int (*task)(void *);
1713 static int sync_task(void *data)
1715 struct sync_task_data *std = data;
1716 std->fail = std->task(std->task_data);
1718 ast_mutex_lock(&std->lock);
1720 ast_cond_signal(&std->cond);
1721 ast_mutex_unlock(&std->lock);
1725 int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1727 /* This method is an onion */
1728 struct sync_task_data std;
1729 ast_mutex_init(&std.lock);
1730 ast_cond_init(&std.cond, NULL);
1731 std.fail = std.complete = 0;
1732 std.task = sip_task;
1733 std.task_data = task_data;
1736 if (ast_taskprocessor_push(serializer, sync_task, &std)) {
1740 if (ast_threadpool_push(sip_threadpool, sync_task, &std)) {
1745 ast_mutex_lock(&std.lock);
1746 while (!std.complete) {
1747 ast_cond_wait(&std.cond, &std.lock);
1749 ast_mutex_unlock(&std.lock);
1751 ast_mutex_destroy(&std.lock);
1752 ast_cond_destroy(&std.cond);
1756 void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
1758 size_t chars_to_copy = MIN(size - 1, pj_strlen(src));
1759 memcpy(dest, pj_strbuf(src), chars_to_copy);
1760 dest[chars_to_copy] = '\0';
1763 int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype)
1765 pjsip_media_type compare;
1767 if (!content_type) {
1771 pjsip_media_type_init2(&compare, type, subtype);
1773 return pjsip_media_type_cmp(content_type, &compare, 0) ? 0 : -1;
1776 pj_caching_pool caching_pool;
1777 pj_pool_t *memory_pool;
1778 pj_thread_t *monitor_thread;
1779 static int monitor_continue;
1781 static void *monitor_thread_exec(void *endpt)
1783 while (monitor_continue) {
1784 const pj_time_val delay = {0, 10};
1785 pjsip_endpt_handle_events(ast_pjsip_endpoint, &delay);
1790 static void stop_monitor_thread(void)
1792 monitor_continue = 0;
1793 pj_thread_join(monitor_thread);
1796 AST_THREADSTORAGE(pj_thread_storage);
1797 AST_THREADSTORAGE(servant_id_storage);
1798 #define SIP_SERVANT_ID 0x5E2F1D
1800 static void sip_thread_start(void)
1802 pj_thread_desc *desc;
1803 pj_thread_t *thread;
1804 uint32_t *servant_id;
1806 servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1808 ast_log(LOG_ERROR, "Could not set SIP servant ID in thread-local storage.\n");
1811 *servant_id = SIP_SERVANT_ID;
1813 desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc));
1815 ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n");
1818 pj_bzero(*desc, sizeof(*desc));
1820 if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) {
1821 ast_log(LOG_ERROR, "Couldn't register thread with PJLIB.\n");
1825 int ast_sip_thread_is_servant(void)
1827 uint32_t *servant_id;
1829 servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1834 return *servant_id == SIP_SERVANT_ID;
1837 static void remove_request_headers(pjsip_endpoint *endpt)
1839 const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt);
1840 pjsip_hdr *iter = request_headers->next;
1842 while (iter != request_headers) {
1843 pjsip_hdr *to_erase = iter;
1845 pj_list_erase(to_erase);
1849 static int load_module(void)
1851 /* The third parameter is just copied from
1852 * example code from PJLIB. This can be adjusted
1856 struct ast_threadpool_options options;
1858 if (pj_init() != PJ_SUCCESS) {
1859 return AST_MODULE_LOAD_DECLINE;
1862 if (pjlib_util_init() != PJ_SUCCESS) {
1864 return AST_MODULE_LOAD_DECLINE;
1867 pj_caching_pool_init(&caching_pool, NULL, 1024 * 1024);
1868 if (pjsip_endpt_create(&caching_pool.factory, "SIP", &ast_pjsip_endpoint) != PJ_SUCCESS) {
1869 ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n");
1870 pj_caching_pool_destroy(&caching_pool);
1871 return AST_MODULE_LOAD_DECLINE;
1874 /* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that,
1875 * we need to stop PJSIP from doing it automatically
1877 remove_request_headers(ast_pjsip_endpoint);
1879 memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL);
1881 ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n");
1882 pjsip_endpt_destroy(ast_pjsip_endpoint);
1883 ast_pjsip_endpoint = NULL;
1884 pj_caching_pool_destroy(&caching_pool);
1885 return AST_MODULE_LOAD_DECLINE;
1888 if (ast_sip_initialize_system()) {
1889 ast_log(LOG_ERROR, "Failed to initialize SIP system configuration. Aborting load\n");
1890 pj_pool_release(memory_pool);
1892 pjsip_endpt_destroy(ast_pjsip_endpoint);
1893 ast_pjsip_endpoint = NULL;
1894 pj_caching_pool_destroy(&caching_pool);
1895 return AST_MODULE_LOAD_DECLINE;
1898 sip_get_threadpool_options(&options);
1899 options.thread_start = sip_thread_start;
1900 sip_threadpool = ast_threadpool_create("SIP", NULL, &options);
1901 if (!sip_threadpool) {
1902 ast_log(LOG_ERROR, "Failed to create SIP threadpool. Aborting load\n");
1903 pj_pool_release(memory_pool);
1905 pjsip_endpt_destroy(ast_pjsip_endpoint);
1906 ast_pjsip_endpoint = NULL;
1907 pj_caching_pool_destroy(&caching_pool);
1908 return AST_MODULE_LOAD_DECLINE;
1911 pjsip_tsx_layer_init_module(ast_pjsip_endpoint);
1912 pjsip_ua_init_module(ast_pjsip_endpoint, NULL);
1914 monitor_continue = 1;
1915 status = pj_thread_create(memory_pool, "SIP", (pj_thread_proc *) &monitor_thread_exec,
1916 NULL, PJ_THREAD_DEFAULT_STACK_SIZE * 2, 0, &monitor_thread);
1917 if (status != PJ_SUCCESS) {
1918 ast_log(LOG_ERROR, "Failed to start SIP monitor thread. Aborting load\n");
1919 pj_pool_release(memory_pool);
1921 pjsip_endpt_destroy(ast_pjsip_endpoint);
1922 ast_pjsip_endpoint = NULL;
1923 pj_caching_pool_destroy(&caching_pool);
1924 return AST_MODULE_LOAD_DECLINE;
1927 ast_sip_initialize_global_headers();
1929 if (ast_res_pjsip_initialize_configuration()) {
1930 ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
1931 ast_sip_destroy_global_headers();
1932 stop_monitor_thread();
1933 pj_pool_release(memory_pool);
1935 pjsip_endpt_destroy(ast_pjsip_endpoint);
1936 ast_pjsip_endpoint = NULL;
1937 pj_caching_pool_destroy(&caching_pool);
1938 return AST_MODULE_LOAD_DECLINE;
1941 if (ast_sip_initialize_distributor()) {
1942 ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n");
1943 ast_res_pjsip_destroy_configuration();
1944 ast_sip_destroy_global_headers();
1945 stop_monitor_thread();
1946 pj_pool_release(memory_pool);
1948 pjsip_endpt_destroy(ast_pjsip_endpoint);
1949 ast_pjsip_endpoint = NULL;
1950 pj_caching_pool_destroy(&caching_pool);
1951 return AST_MODULE_LOAD_DECLINE;
1954 if (ast_sip_initialize_outbound_authentication()) {
1955 ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n");
1956 ast_sip_destroy_distributor();
1957 ast_res_pjsip_destroy_configuration();
1958 ast_sip_destroy_global_headers();
1959 stop_monitor_thread();
1960 pj_pool_release(memory_pool);
1962 pjsip_endpt_destroy(ast_pjsip_endpoint);
1963 ast_pjsip_endpoint = NULL;
1964 pj_caching_pool_destroy(&caching_pool);
1965 return AST_MODULE_LOAD_DECLINE;
1968 ast_res_pjsip_init_options_handling(0);
1970 ast_res_pjsip_init_contact_transports();
1972 ast_module_ref(ast_module_info->self);
1974 return AST_MODULE_LOAD_SUCCESS;
1977 static int reload_module(void)
1979 if (ast_res_pjsip_reload_configuration()) {
1980 return AST_MODULE_LOAD_DECLINE;
1982 ast_res_pjsip_init_options_handling(1);
1986 static int unload_module(void)
1988 /* This will never get called as this module can't be unloaded */
1992 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Basic SIP resource",
1993 .load = load_module,
1994 .unload = unload_module,
1995 .reload = reload_module,
1996 .load_pri = AST_MODPRI_CHANNEL_DEPEND - 5,
projects / asterisk/asterisk.git / blob