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 There are currently two methods to identify an endpoint. By default
224 both are used to identify an endpoint.
227 <enum name="username" />
228 <enum name="location" />
229 <enum name="username,location" />
233 <configOption name="mailboxes">
234 <synopsis>Mailbox(es) to be associated with</synopsis>
236 <configOption name="mohsuggest" default="default">
237 <synopsis>Default Music On Hold class</synopsis>
239 <configOption name="outbound_auth">
240 <synopsis>Authentication object used for outbound requests</synopsis>
242 <configOption name="outbound_proxy">
243 <synopsis>Proxy through which to send requests</synopsis>
245 <configOption name="rewrite_contact">
246 <synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
248 <configOption name="rtp_ipv6" default="no">
249 <synopsis>Allow use of IPv6 for RTP traffic</synopsis>
251 <configOption name="rtp_symmetric" default="no">
252 <synopsis>Enforce that RTP must be symmetric</synopsis>
254 <configOption name="send_pai" default="no">
255 <synopsis>Send the P-Asserted-Identity header</synopsis>
257 <configOption name="send_rpid" default="no">
258 <synopsis>Send the Remote-Party-ID header</synopsis>
260 <configOption name="timers_min_se" default="90">
261 <synopsis>Minimum session timers expiration period</synopsis>
263 Minimium session timer expiration period. Time in seconds.
264 </para></description>
266 <configOption name="timers" default="yes">
267 <synopsis>Session timers for SIP packets</synopsis>
270 <enum name="forced" />
272 <enum name="required" />
277 <configOption name="timers_sess_expires" default="1800">
278 <synopsis>Maximum session timer expiration period</synopsis>
280 Maximium session timer expiration period. Time in seconds.
281 </para></description>
283 <configOption name="transport">
284 <synopsis>Desired transport configuration</synopsis>
286 This will set the desired transport configuration to send SIP data through.
288 <warning><para>Not specifying a transport will <emphasis>DEFAULT</emphasis>
289 to the first configured transport in <filename>pjsip.conf</filename> which is
290 valid for the URI we are trying to contact.
294 <configOption name="trust_id_inbound" default="no">
295 <synopsis>Accept identification information received from this endpoint</synopsis>
296 <description><para>This option determines whether Asterisk will accept
297 identification from the endpoint from headers such as P-Asserted-Identity
298 or Remote-Party-ID header. This option applies both to calls originating from the
299 endpoint and calls originating from Asterisk. If <literal>no</literal>, the
300 configured Caller-ID from pjsip.conf will always be used as the identity for
301 the endpoint.</para></description>
303 <configOption name="trust_id_outbound" default="no">
304 <synopsis>Send private identification details to the endpoint.</synopsis>
305 <description><para>This option determines whether res_pjsip will send private
306 identification information to the endpoint. If <literal>no</literal>,
307 private Caller-ID information will not be forwarded to the endpoint.
308 "Private" in this case refers to any method of restricting identification.
309 Example: setting <replaceable>callerid_privacy</replaceable> to any
310 <literal>prohib</literal> variation.
311 Example: If <replaceable>trust_id_inbound</replaceable> is set to
312 <literal>yes</literal>, the presence of a <literal>Privacy: id</literal>
313 header in a SIP request or response would indicate the identification
314 provided in the request is private.</para></description>
316 <configOption name="type">
317 <synopsis>Must be of type 'endpoint'.</synopsis>
319 <configOption name="use_ptime" default="no">
320 <synopsis>Use Endpoint's requested packetisation interval</synopsis>
322 <configOption name="use_avpf" default="no">
323 <synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this
326 If set to <literal>yes</literal>, res_pjsip will use use the AVPF or SAVPF RTP
327 profile for all media offers on outbound calls and media updates and will
328 decline media offers not using the AVPF or SAVPF profile.
330 If set to <literal>no</literal>, res_pjsip will use use the AVP or SAVP RTP
331 profile for all media offers on outbound calls and media updates and will
332 decline media offers not using the AVP or SAVP profile.
333 </para></description>
335 <configOption name="media_encryption" default="no">
336 <synopsis>Determines whether res_pjsip will use and enforce usage of media encryption
337 for this endpoint.</synopsis>
340 <enum name="no"><para>
341 res_pjsip will offer no encryption and allow no encryption to be setup.
343 <enum name="sdes"><para>
344 res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP
345 transport should be used in conjunction with this option to prevent
346 exposure of media encryption keys.
348 <enum name="dtls"><para>
349 res_pjsip will offer DTLS-SRTP setup.
354 <configOption name="inband_progress" default="no">
355 <synopsis>Determines whether chan_pjsip will indicate ringing using inband
358 If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
359 when told to indicate ringing and will immediately start sending ringing
362 If set to <literal>no</literal>, chan_pjsip will send a 180 Ringing when told
363 to indicate ringing and will NOT send it as audio.
364 </para></description>
366 <configOption name="callgroup">
367 <synopsis>The numeric pickup groups for a channel.</synopsis>
369 Can be set to a comma separated list of numbers or ranges between the values
370 of 0-63 (maximum of 64 groups).
371 </para></description>
373 <configOption name="pickupgroup">
374 <synopsis>The numeric pickup groups that a channel can pickup.</synopsis>
376 Can be set to a comma separated list of numbers or ranges between the values
377 of 0-63 (maximum of 64 groups).
378 </para></description>
380 <configOption name="namedcallgroup">
381 <synopsis>The named pickup groups for a channel.</synopsis>
383 Can be set to a comma separated list of case sensitive strings limited by
384 supported line length.
385 </para></description>
387 <configOption name="namedpickupgroup">
388 <synopsis>The named pickup groups that a channel can pickup.</synopsis>
390 Can be set to a comma separated list of case sensitive strings limited by
391 supported line length.
392 </para></description>
394 <configOption name="devicestate_busy_at" default="0">
395 <synopsis>The number of in-use channels which will cause busy to be returned as device state</synopsis>
397 When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the
398 PJSIP channel driver will return busy as the device state instead of in use.
399 </para></description>
401 <configOption name="t38udptl" default="no">
402 <synopsis>Whether T.38 UDPTL support is enabled or not</synopsis>
404 If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted
406 </para></description>
408 <configOption name="t38udptl_ec" default="none">
409 <synopsis>T.38 UDPTL error correction method</synopsis>
412 <enum name="none"><para>
413 No error correction should be used.
415 <enum name="fec"><para>
416 Forward error correction should be used.
418 <enum name="redundancy"><para>
419 Redundacy error correction should be used.
424 <configOption name="t38udptl_maxdatagram" default="0">
425 <synopsis>T.38 UDPTL maximum datagram size</synopsis>
427 This option can be set to override the maximum datagram of a remote endpoint for broken
429 </para></description>
431 <configOption name="faxdetect" default="no">
432 <synopsis>Whether CNG tone detection is enabled</synopsis>
434 This option can be set to send the session to the fax extension when a CNG tone is
436 </para></description>
438 <configOption name="t38udptl_nat" default="no">
439 <synopsis>Whether NAT support is enabled on UDPTL sessions</synopsis>
441 When enabled the UDPTL stack will send UDPTL packets to the source address of
443 </para></description>
445 <configOption name="t38udptl_ipv6" default="no">
446 <synopsis>Whether IPv6 is used for UDPTL Sessions</synopsis>
448 When enabled the UDPTL stack will use IPv6.
449 </para></description>
451 <configOption name="tonezone">
452 <synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis>
454 <configOption name="language">
455 <synopsis>Set the default language to use for channels created for this endpoint.</synopsis>
457 <configOption name="one_touch_recording" default="no">
458 <synopsis>Determines whether one-touch recording is allowed for this endpoint.</synopsis>
460 <ref type="configOption">recordonfeature</ref>
461 <ref type="configOption">recordofffeature</ref>
464 <configOption name="recordonfeature" default="automixmon">
465 <synopsis>The feature to enact when one-touch recording is turned on.</synopsis>
467 <para>When an INFO request for one-touch recording arrives with a Record header set to "on", this
468 feature will be enabled for the channel. The feature designated here can be any built-in
469 or dynamic feature defined in features.conf.</para>
470 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
473 <ref type="configOption">one_touch_recording</ref>
474 <ref type="configOption">recordofffeature</ref>
477 <configOption name="recordofffeature" default="automixmon">
478 <synopsis>The feature to enact when one-touch recording is turned off.</synopsis>
480 <para>When an INFO request for one-touch recording arrives with a Record header set to "off", this
481 feature will be enabled for the channel. The feature designated here can be any built-in
482 or dynamic feature defined in features.conf.</para>
483 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
486 <ref type="configOption">one_touch_recording</ref>
487 <ref type="configOption">recordonfeature</ref>
490 <configOption name="rtpengine" default="asterisk">
491 <synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis>
493 <configOption name="allowtransfer" default="yes">
494 <synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis>
496 <configOption name="sdpowner" default="-">
497 <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
499 <configOption name="sdpsession" default="Asterisk">
500 <synopsis>String used for the SDP session (s=) line.</synopsis>
502 <configOption name="tos_audio">
503 <synopsis>DSCP TOS bits for audio streams</synopsis>
505 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
506 </para></description>
508 <configOption name="tos_video">
509 <synopsis>DSCP TOS bits for video streams</synopsis>
511 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
512 </para></description>
514 <configOption name="cos_audio">
515 <synopsis>Priority for audio streams</synopsis>
517 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
518 </para></description>
520 <configOption name="cos_video">
521 <synopsis>Priority for video streams</synopsis>
523 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
524 </para></description>
526 <configOption name="allowsubscribe" default="yes">
527 <synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis>
529 <configOption name="subminexpiry" default="60">
530 <synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis>
532 <configOption name="fromuser">
533 <synopsis>Username to use in From header for requests to this endpoint.</synopsis>
535 <configOption name="mwifromuser">
536 <synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis>
538 <configOption name="fromdomain">
539 <synopsis>Domain to user in From header for requests to this endpoint.</synopsis>
541 <configOption name="dtlsverify">
542 <synopsis>Verify that the provided peer certificate is valid</synopsis>
544 This option only applies if <replaceable>media_encryption</replaceable> is
545 set to <literal>dtls</literal>.
546 </para></description>
548 <configOption name="dtlsrekey">
549 <synopsis>Interval at which to renegotiate the TLS session and rekey the SRTP session</synopsis>
551 This option only applies if <replaceable>media_encryption</replaceable> is
552 set to <literal>dtls</literal>.
554 If this is not set or the value provided is 0 rekeying will be disabled.
555 </para></description>
557 <configOption name="dtlscertfile">
558 <synopsis>Path to certificate file to present to peer</synopsis>
560 This option only applies if <replaceable>media_encryption</replaceable> is
561 set to <literal>dtls</literal>.
562 </para></description>
564 <configOption name="dtlsprivatekey">
565 <synopsis>Path to private key for certificate file</synopsis>
567 This option only applies if <replaceable>media_encryption</replaceable> is
568 set to <literal>dtls</literal>.
569 </para></description>
571 <configOption name="dtlscipher">
572 <synopsis>Cipher to use for DTLS negotiation</synopsis>
574 This option only applies if <replaceable>media_encryption</replaceable> is
575 set to <literal>dtls</literal>.
577 Many options for acceptable ciphers. See link for more:
578 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
579 </para></description>
581 <configOption name="dtlscafile">
582 <synopsis>Path to certificate authority certificate</synopsis>
584 This option only applies if <replaceable>media_encryption</replaceable> is
585 set to <literal>dtls</literal>.
586 </para></description>
588 <configOption name="dtlscapath">
589 <synopsis>Path to a directory containing certificate authority certificates</synopsis>
591 This option only applies if <replaceable>media_encryption</replaceable> is
592 set to <literal>dtls</literal>.
593 </para></description>
595 <configOption name="dtlssetup">
596 <synopsis>Whether we are willing to accept connections, connect to the other party, or both.</synopsis>
599 This option only applies if <replaceable>media_encryption</replaceable> is
600 set to <literal>dtls</literal>.
603 <enum name="active"><para>
604 res_pjsip will make a connection to the peer.
606 <enum name="passive"><para>
607 res_pjsip will accept connections from the peer.
609 <enum name="actpass"><para>
610 res_pjsip will offer and accept connections from the peer.
615 <configOption name="srtp_tag_32">
616 <synopsis>Determines whether 32 byte tags should be used instead of 80 byte tags.</synopsis>
618 This option only applies if <replaceable>media_encryption</replaceable> is
619 set to <literal>sdes</literal> or <literal>dtls</literal>.
620 </para></description>
623 <configObject name="auth">
624 <synopsis>Authentication type</synopsis>
626 Authentication objects hold the authenitcation information for use
627 by <literal>endpoints</literal>. This also allows for multiple <literal>
628 endpoints</literal> to use the same information. Choice of MD5/plaintext
629 and setting of username.
630 </para></description>
631 <configOption name="auth_type" default="userpass">
632 <synopsis>Authentication type</synopsis>
634 This option specifies which of the password style config options should be read,
635 either 'password' or 'md5_cred' when trying to authenticate an endpoint inbound request.
639 <enum name="userpass"/>
643 <configOption name="nonce_lifetime" default="32">
644 <synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
646 <configOption name="md5_cred">
647 <synopsis>MD5 Hash used for authentication.</synopsis>
648 <description><para>Only used when auth_type is <literal>md5</literal>.</para></description>
650 <configOption name="password">
651 <synopsis>PlainText password used for authentication.</synopsis>
652 <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
654 <configOption name="realm" default="asterisk">
655 <synopsis>SIP realm for endpoint</synopsis>
657 <configOption name="type">
658 <synopsis>Must be 'auth'</synopsis>
660 <configOption name="username">
661 <synopsis>Username to use for account</synopsis>
664 <configObject name="nat_hook">
665 <synopsis>XXX This exists only to prevent XML documentation errors.</synopsis>
666 <configOption name="external_media_address">
667 <synopsis>I should be undocumented or hidden</synopsis>
669 <configOption name="method">
670 <synopsis>I should be undocumented or hidden</synopsis>
673 <configObject name="domain_alias">
674 <synopsis>Domain Alias</synopsis>
676 Signifies that a domain is an alias. Used for checking the domain of
677 the AoR to which the endpoint is binding.
678 </para></description>
679 <configOption name="type">
680 <synopsis>Must be of type 'domain_alias'.</synopsis>
682 <configOption name="domain">
683 <synopsis>Domain to be aliased</synopsis>
686 <configObject name="transport">
687 <synopsis>SIP Transport</synopsis>
689 <emphasis>Transports</emphasis>
691 <para>There are different transports and protocol derivatives
692 supported by <literal>res_pjsip</literal>. They are in order of
693 preference: UDP, TCP, and WebSocket (WS).</para>
695 Multiple endpoints using the same connection is <emphasis>NOT</emphasis>
696 supported. Doing so may result in broken calls.
699 <configOption name="async_operations" default="1">
700 <synopsis>Number of simultaneous Asynchronous Operations</synopsis>
702 <configOption name="bind">
703 <synopsis>IP Address and optional port to bind to for this transport</synopsis>
705 <configOption name="ca_list_file">
706 <synopsis>File containing a list of certificates to read (TLS ONLY)</synopsis>
708 <configOption name="cert_file">
709 <synopsis>Certificate file for endpoint (TLS ONLY)</synopsis>
711 <configOption name="cipher">
712 <synopsis>Preferred Cryptography Cipher (TLS ONLY)</synopsis>
714 Many options for acceptable ciphers see link for more:
715 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
716 </para></description>
718 <configOption name="domain">
719 <synopsis>Domain the transport comes from</synopsis>
721 <configOption name="external_media_address">
722 <synopsis>External Address to use in RTP handling</synopsis>
724 <configOption name="external_signaling_address">
725 <synopsis>External address for SIP signalling</synopsis>
727 <configOption name="external_signaling_port" default="0">
728 <synopsis>External port for SIP signalling</synopsis>
730 <configOption name="method">
731 <synopsis>Method of SSL transport (TLS ONLY)</synopsis>
734 <enum name="default" />
735 <enum name="unspecified" />
736 <enum name="tlsv1" />
737 <enum name="sslv2" />
738 <enum name="sslv3" />
739 <enum name="sslv23" />
743 <configOption name="localnet">
744 <synopsis>Network to consider local (used for NAT purposes).</synopsis>
745 <description><para>This must be in CIDR or dotted decimal format with the IP
746 and mask separated with a slash ('/').</para></description>
748 <configOption name="password">
749 <synopsis>Password required for transport</synopsis>
751 <configOption name="privkey_file">
752 <synopsis>Private key file (TLS ONLY)</synopsis>
754 <configOption name="protocol" default="udp">
755 <synopsis>Protocol to use for SIP traffic</synopsis>
764 <configOption name="require_client_cert" default="false">
765 <synopsis>Require client certificate (TLS ONLY)</synopsis>
767 <configOption name="type">
768 <synopsis>Must be of type 'transport'.</synopsis>
770 <configOption name="verify_client" default="false">
771 <synopsis>Require verification of client certificate (TLS ONLY)</synopsis>
773 <configOption name="verify_server" default="false">
774 <synopsis>Require verification of server certificate (TLS ONLY)</synopsis>
777 <configObject name="contact">
778 <synopsis>A way of creating an aliased name to a SIP URI</synopsis>
780 Contacts are a way to hide SIP URIs from the dialplan directly.
781 They are also used to make a group of contactable parties when
782 in use with <literal>AoR</literal> lists.
783 </para></description>
784 <configOption name="type">
785 <synopsis>Must be of type 'contact'.</synopsis>
787 <configOption name="uri">
788 <synopsis>SIP URI to contact peer</synopsis>
790 <configOption name="expiration_time">
791 <synopsis>Time to keep alive a contact</synopsis>
793 Time to keep alive a contact. String style specification.
794 </para></description>
796 <configOption name="qualify_frequency" default="0">
797 <synopsis>Interval at which to qualify a contact</synopsis>
799 Interval between attempts to qualify the contact for reachability.
800 If <literal>0</literal> never qualify. Time in seconds.
801 </para></description>
804 <configObject name="contact_status">
805 <synopsis>Status for a contact</synopsis>
807 The contact status keeps track of whether or not a contact is reachable
808 and how long it took to qualify the contact (round trip time).
809 </para></description>
810 <configOption name="status">
811 <synopsis>A contact's status</synopsis>
814 <enum name="AVAILABLE" />
815 <enum name="UNAVAILABLE" />
819 <configOption name="rtt">
820 <synopsis>Round trip time</synopsis>
822 The time, in microseconds, it took to qualify the contact.
823 </para></description>
826 <configObject name="aor">
827 <synopsis>The configuration for a location of an endpoint</synopsis>
829 An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
830 AoRs are specified, an endpoint will not be reachable by Asterisk.
831 Beyond that, an AoR has other uses within Asterisk.
833 An <literal>AoR</literal> is a way to allow dialing a group
834 of <literal>Contacts</literal> that all use the same
835 <literal>endpoint</literal> for calls.
837 This can be used as another way of grouping a list of contacts to dial
838 rather than specifing them each directly when dialing via the dialplan.
839 This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
840 </para></description>
841 <configOption name="contact">
842 <synopsis>Permanent contacts assigned to AoR</synopsis>
844 Contacts included in this list will be called whenever referenced
845 by <literal>chan_pjsip</literal>.
846 </para></description>
848 <configOption name="default_expiration" default="3600">
849 <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
851 <configOption name="mailboxes">
852 <synopsis>Mailbox(es) to be associated with</synopsis>
853 <description><para>This option applies when an external entity subscribes to an AoR
854 for message waiting indications. The mailboxes specified here will be
855 subscribed to.</para></description>
857 <configOption name="maximum_expiration" default="7200">
858 <synopsis>Maximum time to keep an AoR</synopsis>
860 Maximium time to keep a peer with explicit expiration. Time in seconds.
861 </para></description>
863 <configOption name="max_contacts" default="0">
864 <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
866 Maximum number of contacts that can associate with this AoR.
868 <note><para>This should be set to <literal>1</literal> and
869 <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
870 wish to stick with the older <literal>chan_sip</literal> behaviour.
874 <configOption name="minimum_expiration" default="60">
875 <synopsis>Minimum keep alive time for an AoR</synopsis>
877 Minimum time to keep a peer with an explict expiration. Time in seconds.
878 </para></description>
880 <configOption name="remove_existing" default="no">
881 <synopsis>Determines whether new contacts replace existing ones.</synopsis>
883 On receiving a new registration to the AoR should it remove
884 the existing contact that was registered against it?
886 <note><para>This should be set to <literal>yes</literal> and
887 <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
888 wish to stick with the older <literal>chan_sip</literal> behaviour.
892 <configOption name="type">
893 <synopsis>Must be of type 'aor'.</synopsis>
895 <configOption name="qualify_frequency" default="0">
896 <synopsis>Interval at which to qualify an AoR</synopsis>
898 Interval between attempts to qualify the AoR for reachability.
899 If <literal>0</literal> never qualify. Time in seconds.
900 </para></description>
902 <configOption name="authenticate_qualify" default="no">
903 <synopsis>Authenticates a qualify request if needed</synopsis>
905 If true and a qualify request receives a challenge or authenticate response
906 authentication is attempted before declaring the contact available.
907 </para></description>
910 <configObject name="system">
911 <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
913 The settings in this section are global. In addition to being global, the values will
914 not be re-evaluated when a reload is performed. This is because the values must be set
915 before the SIP stack is initialized. The only way to reset these values is to either
916 restart Asterisk, or unload res_pjsip.so and then load it again.
917 </para></description>
918 <configOption name="timert1" default="500">
919 <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
921 Timer T1 is the base for determining how long to wait before retransmitting
922 requests that receive no response when using an unreliable transport (e.g. UDP).
923 For more information on this timer, see RFC 3261, Section 17.1.1.1.
924 </para></description>
926 <configOption name="timerb" default="32000">
927 <synopsis>Set transaction timer B value (milliseconds).</synopsis>
929 Timer B determines the maximum amount of time to wait after sending an INVITE
930 request before terminating the transaction. It is recommended that this be set
931 to 64 * Timer T1, but it may be set higher if desired. For more information on
932 this timer, see RFC 3261, Section 17.1.1.1.
933 </para></description>
935 <configOption name="compactheaders" default="no">
936 <synopsis>Use the short forms of common SIP header names.</synopsis>
939 <configObject name="global">
940 <synopsis>Options that apply globally to all SIP communications</synopsis>
942 The settings in this section are global. Unlike options in the <literal>system</literal>
943 section, these options can be refreshed by performing a reload.
944 </para></description>
945 <configOption name="maxforwards" default="70">
946 <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
948 <configOption name="useragent" default="Asterisk <Asterisk Version>">
949 <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
954 <manager name="PJSIPQualify" language="en_US">
956 Qualify a chan_pjsip endpoint.
959 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
960 <parameter name="Endpoint" required="true">
961 <para>The endpoint you want to qualify.</para>
965 <para>Qualify a chan_pjsip endpoint.</para>
971 static pjsip_endpoint *ast_pjsip_endpoint;
973 static struct ast_threadpool *sip_threadpool;
975 static int register_service(void *data)
977 pjsip_module **module = data;
978 if (!ast_pjsip_endpoint) {
979 ast_log(LOG_ERROR, "There is no PJSIP endpoint. Unable to register services\n");
982 if (pjsip_endpt_register_module(ast_pjsip_endpoint, *module) != PJ_SUCCESS) {
983 ast_log(LOG_ERROR, "Unable to register module %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
986 ast_debug(1, "Registered SIP service %.*s (%p)\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name), *module);
987 ast_module_ref(ast_module_info->self);
991 int ast_sip_register_service(pjsip_module *module)
993 return ast_sip_push_task_synchronous(NULL, register_service, &module);
996 static int unregister_service(void *data)
998 pjsip_module **module = data;
999 ast_module_unref(ast_module_info->self);
1000 if (!ast_pjsip_endpoint) {
1003 pjsip_endpt_unregister_module(ast_pjsip_endpoint, *module);
1004 ast_debug(1, "Unregistered SIP service %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1008 void ast_sip_unregister_service(pjsip_module *module)
1010 ast_sip_push_task_synchronous(NULL, unregister_service, &module);
1013 static struct ast_sip_authenticator *registered_authenticator;
1015 int ast_sip_register_authenticator(struct ast_sip_authenticator *auth)
1017 if (registered_authenticator) {
1018 ast_log(LOG_WARNING, "Authenticator %p is already registered. Cannot register a new one\n", registered_authenticator);
1021 registered_authenticator = auth;
1022 ast_debug(1, "Registered SIP authenticator module %p\n", auth);
1023 ast_module_ref(ast_module_info->self);
1027 void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth)
1029 if (registered_authenticator != auth) {
1030 ast_log(LOG_WARNING, "Trying to unregister authenticator %p but authenticator %p registered\n",
1031 auth, registered_authenticator);
1034 registered_authenticator = NULL;
1035 ast_debug(1, "Unregistered SIP authenticator %p\n", auth);
1036 ast_module_unref(ast_module_info->self);
1039 int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1041 if (!registered_authenticator) {
1042 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is not required\n");
1046 return registered_authenticator->requires_authentication(endpoint, rdata);
1049 enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
1050 pjsip_rx_data *rdata, pjsip_tx_data *tdata)
1052 if (!registered_authenticator) {
1053 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is successful\n");
1056 return registered_authenticator->check_authentication(endpoint, rdata, tdata);
1059 static struct ast_sip_outbound_authenticator *registered_outbound_authenticator;
1061 int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1063 if (registered_outbound_authenticator) {
1064 ast_log(LOG_WARNING, "Outbound authenticator %p is already registered. Cannot register a new one\n", registered_outbound_authenticator);
1067 registered_outbound_authenticator = auth;
1068 ast_debug(1, "Registered SIP outbound authenticator module %p\n", auth);
1069 ast_module_ref(ast_module_info->self);
1073 void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1075 if (registered_outbound_authenticator != auth) {
1076 ast_log(LOG_WARNING, "Trying to unregister outbound authenticator %p but outbound authenticator %p registered\n",
1077 auth, registered_outbound_authenticator);
1080 registered_outbound_authenticator = NULL;
1081 ast_debug(1, "Unregistered SIP outbound authenticator %p\n", auth);
1082 ast_module_unref(ast_module_info->self);
1085 int ast_sip_create_request_with_auth(const struct ast_sip_auth_array *auths, pjsip_rx_data *challenge,
1086 pjsip_transaction *tsx, pjsip_tx_data **new_request)
1088 if (!registered_outbound_authenticator) {
1089 ast_log(LOG_WARNING, "No SIP outbound authenticator registered. Cannot respond to authentication challenge\n");
1092 return registered_outbound_authenticator->create_request_with_auth(auths, challenge, tsx, new_request);
1095 struct endpoint_identifier_list {
1096 struct ast_sip_endpoint_identifier *identifier;
1097 AST_RWLIST_ENTRY(endpoint_identifier_list) list;
1100 static AST_RWLIST_HEAD_STATIC(endpoint_identifiers, endpoint_identifier_list);
1102 int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1104 struct endpoint_identifier_list *id_list_item;
1105 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1107 id_list_item = ast_calloc(1, sizeof(*id_list_item));
1108 if (!id_list_item) {
1109 ast_log(LOG_ERROR, "Unabled to add endpoint identifier. Out of memory.\n");
1112 id_list_item->identifier = identifier;
1114 AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list);
1115 ast_debug(1, "Registered endpoint identifier %p\n", identifier);
1117 ast_module_ref(ast_module_info->self);
1121 void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1123 struct endpoint_identifier_list *iter;
1124 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1125 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) {
1126 if (iter->identifier == identifier) {
1127 AST_RWLIST_REMOVE_CURRENT(list);
1129 ast_debug(1, "Unregistered endpoint identifier %p\n", identifier);
1130 ast_module_unref(ast_module_info->self);
1134 AST_RWLIST_TRAVERSE_SAFE_END;
1137 struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata)
1139 struct endpoint_identifier_list *iter;
1140 struct ast_sip_endpoint *endpoint = NULL;
1141 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1142 AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) {
1143 ast_assert(iter->identifier->identify_endpoint != NULL);
1144 endpoint = iter->identifier->identify_endpoint(rdata);
1152 pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
1154 return ast_pjsip_endpoint;
1157 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)
1159 pj_str_t tmp, local_addr;
1161 pjsip_sip_uri *sip_uri;
1162 pjsip_transport_type_e type = PJSIP_TRANSPORT_UNSPECIFIED;
1164 char uuid_str[AST_UUID_STR_LEN];
1166 if (ast_strlen_zero(user)) {
1167 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1171 user = ast_uuid_to_str(uuid, uuid_str, sizeof(uuid_str));
1174 /* Parse the provided target URI so we can determine what transport it will end up using */
1175 pj_strdup_with_null(pool, &tmp, target);
1177 if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) ||
1178 (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
1182 sip_uri = pjsip_uri_get_uri(uri);
1184 /* Determine the transport type to use */
1185 if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
1186 type = PJSIP_TRANSPORT_TLS;
1187 } else if (!sip_uri->transport_param.slen) {
1188 type = PJSIP_TRANSPORT_UDP;
1190 type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
1193 if (type == PJSIP_TRANSPORT_UNSPECIFIED) {
1197 /* If the host is IPv6 turn the transport into an IPv6 version */
1198 if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1199 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1202 if (!ast_strlen_zero(domain)) {
1203 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1204 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1206 (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1209 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1210 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1214 /* Get the local bound address for the transport that will be used when communicating with the provided URI */
1215 if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), pool, type, selector,
1216 &local_addr, &local_port) != PJ_SUCCESS) {
1220 /* If IPv6 was specified in the transport, set the proper type */
1221 if (pj_strchr(&local_addr, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1222 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1225 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1226 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1227 "<%s:%s@%s%.*s%s:%d%s%s>",
1228 (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1230 (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1231 (int)local_addr.slen,
1233 (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1235 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1236 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1241 static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
1243 RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
1244 const char *transport_name = endpoint->transport;
1246 if (ast_strlen_zero(transport_name)) {
1250 transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name);
1252 if (!transport || !transport->state) {
1256 if (transport->state->transport) {
1257 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1258 selector->u.transport = transport->state->transport;
1259 } else if (transport->state->factory) {
1260 selector->type = PJSIP_TPSELECTOR_LISTENER;
1261 selector->u.listener = transport->state->factory;
1269 static int sip_get_tpselector_from_uri(const char *uri, pjsip_tpselector *selector)
1271 RAII_VAR(struct ast_sip_contact_transport *, contact_transport, NULL, ao2_cleanup);
1273 contact_transport = ast_sip_location_retrieve_contact_transport_by_uri(uri);
1275 if (!contact_transport) {
1279 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1280 selector->u.transport = contact_transport->transport;
1285 pjsip_dialog *ast_sip_create_dialog(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
1287 pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri;
1288 pjsip_dialog *dlg = NULL;
1289 const char *outbound_proxy = endpoint->outbound_proxy;
1290 pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1291 static const pj_str_t HCONTACT = { "Contact", 7 };
1293 pj_cstr(&remote_uri, uri);
1295 if (pjsip_dlg_create_uac(pjsip_ua_instance(), &local_uri, NULL, &remote_uri, NULL, &dlg) != PJ_SUCCESS) {
1299 if (sip_get_tpselector_from_uri(uri, &selector) && sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1300 pjsip_dlg_terminate(dlg);
1304 if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
1305 pjsip_dlg_terminate(dlg);
1309 /* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
1310 pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
1311 dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
1312 dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
1314 /* If a request user has been specified and we are permitted to change it, do so */
1315 if (!ast_strlen_zero(request_user) && (PJSIP_URI_SCHEME_IS_SIP(dlg->target) || PJSIP_URI_SCHEME_IS_SIPS(dlg->target))) {
1316 pjsip_sip_uri *target = pjsip_uri_get_uri(dlg->target);
1317 pj_strdup2(dlg->pool, &target->user, request_user);
1320 /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
1323 pjsip_dlg_set_transport(dlg, &selector);
1325 if (!ast_strlen_zero(outbound_proxy)) {
1326 pjsip_route_hdr route_set, *route;
1327 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1330 pj_list_init(&route_set);
1332 pj_strdup2_with_null(dlg->pool, &tmp, outbound_proxy);
1333 if (!(route = pjsip_parse_hdr(dlg->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1334 pjsip_dlg_terminate(dlg);
1337 pj_list_push_back(&route_set, route);
1339 pjsip_dlg_set_route_set(dlg, &route_set);
1347 /* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
1348 const pjsip_method pjsip_info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
1349 const pjsip_method pjsip_message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
1353 const pjsip_method *pmethod;
1355 { "INVITE", &pjsip_invite_method },
1356 { "CANCEL", &pjsip_cancel_method },
1357 { "ACK", &pjsip_ack_method },
1358 { "BYE", &pjsip_bye_method },
1359 { "REGISTER", &pjsip_register_method },
1360 { "OPTIONS", &pjsip_options_method },
1361 { "SUBSCRIBE", &pjsip_subscribe_method },
1362 { "NOTIFY", &pjsip_notify_method },
1363 { "PUBLISH", &pjsip_publish_method },
1364 { "INFO", &pjsip_info_method },
1365 { "MESSAGE", &pjsip_message_method },
1368 static const pjsip_method *get_pjsip_method(const char *method)
1371 for (i = 0; i < ARRAY_LEN(methods); ++i) {
1372 if (!strcmp(method, methods[i].method)) {
1373 return methods[i].pmethod;
1379 static int create_in_dialog_request(const pjsip_method *method, struct pjsip_dialog *dlg, pjsip_tx_data **tdata)
1381 if (pjsip_dlg_create_request(dlg, method, -1, tdata) != PJ_SUCCESS) {
1382 ast_log(LOG_WARNING, "Unable to create in-dialog request.\n");
1389 static int create_out_of_dialog_request(const pjsip_method *method, struct ast_sip_endpoint *endpoint,
1390 const char *uri, pjsip_tx_data **tdata)
1392 RAII_VAR(struct ast_sip_contact *, contact, NULL, ao2_cleanup);
1393 pj_str_t remote_uri;
1396 pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1398 if (ast_strlen_zero(uri)) {
1400 ast_log(LOG_ERROR, "An endpoint and/or uri must be specified\n");
1404 contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
1405 if (!contact || ast_strlen_zero(contact->uri)) {
1406 ast_log(LOG_ERROR, "Unable to retrieve contact for endpoint %s\n",
1407 ast_sorcery_object_get_id(endpoint));
1411 pj_cstr(&remote_uri, contact->uri);
1413 pj_cstr(&remote_uri, uri);
1417 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1418 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
1419 ast_sorcery_object_get_id(endpoint));
1424 pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
1427 ast_log(LOG_ERROR, "Unable to create PJLIB memory pool\n");
1431 if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL,
1432 endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
1433 ast_log(LOG_ERROR, "Unable to create From header for %.*s request to endpoint %s\n",
1434 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1435 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1439 if (pjsip_endpt_create_request(ast_sip_get_pjsip_endpoint(), method, &remote_uri,
1440 &from, &remote_uri, &from, NULL, -1, NULL, tdata) != PJ_SUCCESS) {
1441 ast_log(LOG_ERROR, "Unable to create outbound %.*s request to endpoint %s\n",
1442 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1443 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1447 /* We can release this pool since request creation copied all the necessary
1448 * data into the outbound request's pool
1450 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1454 int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
1455 struct ast_sip_endpoint *endpoint, const char *uri,
1456 pjsip_tx_data **tdata)
1458 const pjsip_method *pmethod = get_pjsip_method(method);
1461 ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
1466 return create_in_dialog_request(pmethod, dlg, tdata);
1468 return create_out_of_dialog_request(pmethod, endpoint, uri, tdata);
1472 static int send_in_dialog_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg)
1474 if (pjsip_dlg_send_request(dlg, tdata, -1, NULL) != PJ_SUCCESS) {
1475 ast_log(LOG_WARNING, "Unable to send in-dialog request.\n");
1481 static void send_request_cb(void *token, pjsip_event *e)
1483 RAII_VAR(struct ast_sip_endpoint *, endpoint, token, ao2_cleanup);
1484 pjsip_transaction *tsx = e->body.tsx_state.tsx;
1485 pjsip_rx_data *challenge = e->body.tsx_state.src.rdata;
1486 pjsip_tx_data *tdata;
1488 if (tsx->status_code != 401 && tsx->status_code != 407) {
1492 if (!ast_sip_create_request_with_auth(&endpoint->outbound_auths, challenge, tsx, &tdata)) {
1493 pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, NULL, NULL);
1497 static int send_out_of_dialog_request(pjsip_tx_data *tdata, struct ast_sip_endpoint *endpoint)
1499 ao2_ref(endpoint, +1);
1500 if (pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, endpoint, send_request_cb) != PJ_SUCCESS) {
1501 ast_log(LOG_ERROR, "Error attempting to send outbound %.*s request to endpoint %s\n",
1502 (int) pj_strlen(&tdata->msg->line.req.method.name),
1503 pj_strbuf(&tdata->msg->line.req.method.name),
1504 ast_sorcery_object_get_id(endpoint));
1505 ao2_ref(endpoint, -1);
1512 int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint)
1514 ast_assert(tdata->msg->type == PJSIP_REQUEST_MSG);
1517 return send_in_dialog_request(tdata, dlg);
1519 return send_out_of_dialog_request(tdata, endpoint);
1523 int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value)
1527 pjsip_generic_string_hdr *hdr;
1529 pj_cstr(&hdr_name, name);
1530 pj_cstr(&hdr_value, value);
1532 hdr = pjsip_generic_string_hdr_create(tdata->pool, &hdr_name, &hdr_value);
1534 pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) hdr);
1538 static pjsip_msg_body *ast_body_to_pjsip_body(pj_pool_t *pool, const struct ast_sip_body *body)
1544 pj_cstr(&type, body->type);
1545 pj_cstr(&subtype, body->subtype);
1546 pj_cstr(&body_text, body->body_text);
1548 return pjsip_msg_body_create(pool, &type, &subtype, &body_text);
1551 int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body)
1553 pjsip_msg_body *pjsip_body = ast_body_to_pjsip_body(tdata->pool, body);
1554 tdata->msg->body = pjsip_body;
1558 int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies)
1561 /* NULL for type and subtype automatically creates "multipart/mixed" */
1562 pjsip_msg_body *body = pjsip_multipart_create(tdata->pool, NULL, NULL);
1564 for (i = 0; i < num_bodies; ++i) {
1565 pjsip_multipart_part *part = pjsip_multipart_create_part(tdata->pool);
1566 part->body = ast_body_to_pjsip_body(tdata->pool, bodies[i]);
1567 pjsip_multipart_add_part(tdata->pool, body, part);
1570 tdata->msg->body = body;
1574 int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text)
1576 size_t combined_size = strlen(body_text) + tdata->msg->body->len;
1577 struct ast_str *body_buffer = ast_str_alloca(combined_size);
1579 ast_str_set(&body_buffer, 0, "%.*s%s", (int) tdata->msg->body->len, (char *) tdata->msg->body->data, body_text);
1581 tdata->msg->body->data = pj_pool_alloc(tdata->pool, combined_size);
1582 pj_memcpy(tdata->msg->body->data, ast_str_buffer(body_buffer), combined_size);
1583 tdata->msg->body->len = combined_size;
1588 struct ast_taskprocessor *ast_sip_create_serializer(void)
1590 struct ast_taskprocessor *serializer;
1591 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1592 char name[AST_UUID_STR_LEN];
1598 ast_uuid_to_str(uuid, name, sizeof(name));
1600 serializer = ast_threadpool_serializer(name, sip_threadpool);
1607 int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1610 return ast_taskprocessor_push(serializer, sip_task, task_data);
1612 return ast_threadpool_push(sip_threadpool, sip_task, task_data);
1616 struct sync_task_data {
1621 int (*task)(void *);
1625 static int sync_task(void *data)
1627 struct sync_task_data *std = data;
1628 std->fail = std->task(std->task_data);
1630 ast_mutex_lock(&std->lock);
1632 ast_cond_signal(&std->cond);
1633 ast_mutex_unlock(&std->lock);
1637 int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1639 /* This method is an onion */
1640 struct sync_task_data std;
1641 ast_mutex_init(&std.lock);
1642 ast_cond_init(&std.cond, NULL);
1643 std.fail = std.complete = 0;
1644 std.task = sip_task;
1645 std.task_data = task_data;
1648 if (ast_taskprocessor_push(serializer, sync_task, &std)) {
1652 if (ast_threadpool_push(sip_threadpool, sync_task, &std)) {
1657 ast_mutex_lock(&std.lock);
1658 while (!std.complete) {
1659 ast_cond_wait(&std.cond, &std.lock);
1661 ast_mutex_unlock(&std.lock);
1663 ast_mutex_destroy(&std.lock);
1664 ast_cond_destroy(&std.cond);
1668 void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
1670 size_t chars_to_copy = MIN(size - 1, pj_strlen(src));
1671 memcpy(dest, pj_strbuf(src), chars_to_copy);
1672 dest[chars_to_copy] = '\0';
1675 int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype)
1677 pjsip_media_type compare;
1679 if (!content_type) {
1683 pjsip_media_type_init2(&compare, type, subtype);
1685 return pjsip_media_type_cmp(content_type, &compare, 0) ? -1 : 0;
1688 pj_caching_pool caching_pool;
1689 pj_pool_t *memory_pool;
1690 pj_thread_t *monitor_thread;
1691 static int monitor_continue;
1693 static void *monitor_thread_exec(void *endpt)
1695 while (monitor_continue) {
1696 const pj_time_val delay = {0, 10};
1697 pjsip_endpt_handle_events(ast_pjsip_endpoint, &delay);
1702 static void stop_monitor_thread(void)
1704 monitor_continue = 0;
1705 pj_thread_join(monitor_thread);
1708 AST_THREADSTORAGE(pj_thread_storage);
1709 AST_THREADSTORAGE(servant_id_storage);
1710 #define SIP_SERVANT_ID 0x5E2F1D
1712 static void sip_thread_start(void)
1714 pj_thread_desc *desc;
1715 pj_thread_t *thread;
1716 uint32_t *servant_id;
1718 servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1720 ast_log(LOG_ERROR, "Could not set SIP servant ID in thread-local storage.\n");
1723 *servant_id = SIP_SERVANT_ID;
1725 desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc));
1727 ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n");
1730 pj_bzero(*desc, sizeof(*desc));
1732 if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) {
1733 ast_log(LOG_ERROR, "Couldn't register thread with PJLIB.\n");
1737 int ast_sip_thread_is_servant(void)
1739 uint32_t *servant_id;
1741 servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1746 return *servant_id == SIP_SERVANT_ID;
1749 static void remove_request_headers(pjsip_endpoint *endpt)
1751 const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt);
1752 pjsip_hdr *iter = request_headers->next;
1754 while (iter != request_headers) {
1755 pjsip_hdr *to_erase = iter;
1757 pj_list_erase(to_erase);
1761 static int load_module(void)
1763 /* The third parameter is just copied from
1764 * example code from PJLIB. This can be adjusted
1769 /* XXX For the time being, create hard-coded threadpool
1770 * options. Just bump up by five threads every time we
1771 * don't have any available threads. Idle threads time
1772 * out after a minute. No maximum size
1774 struct ast_threadpool_options options = {
1775 .version = AST_THREADPOOL_OPTIONS_VERSION,
1776 .auto_increment = 5,
1780 .thread_start = sip_thread_start,
1782 sip_threadpool = ast_threadpool_create("SIP", NULL, &options);
1784 if (pj_init() != PJ_SUCCESS) {
1785 return AST_MODULE_LOAD_DECLINE;
1788 if (pjlib_util_init() != PJ_SUCCESS) {
1790 return AST_MODULE_LOAD_DECLINE;
1793 pj_caching_pool_init(&caching_pool, NULL, 1024 * 1024);
1794 if (pjsip_endpt_create(&caching_pool.factory, "SIP", &ast_pjsip_endpoint) != PJ_SUCCESS) {
1795 ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n");
1799 /* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that,
1800 * we need to stop PJSIP from doing it automatically
1802 remove_request_headers(ast_pjsip_endpoint);
1804 memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL);
1806 ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n");
1810 if (ast_sip_initialize_system()) {
1811 ast_log(LOG_ERROR, "Failed to initialize SIP system configuration. Aborting load\n");
1815 pjsip_tsx_layer_init_module(ast_pjsip_endpoint);
1816 pjsip_ua_init_module(ast_pjsip_endpoint, NULL);
1818 monitor_continue = 1;
1819 status = pj_thread_create(memory_pool, "SIP", (pj_thread_proc *) &monitor_thread_exec,
1820 NULL, PJ_THREAD_DEFAULT_STACK_SIZE * 2, 0, &monitor_thread);
1821 if (status != PJ_SUCCESS) {
1822 ast_log(LOG_ERROR, "Failed to start SIP monitor thread. Aborting load\n");
1826 ast_sip_initialize_global_headers();
1828 if (ast_res_pjsip_initialize_configuration()) {
1829 ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
1833 if (ast_sip_initialize_distributor()) {
1834 ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n");
1838 if (ast_sip_initialize_outbound_authentication()) {
1839 ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n");
1843 ast_res_pjsip_init_options_handling(0);
1845 ast_res_pjsip_init_contact_transports();
1847 return AST_MODULE_LOAD_SUCCESS;
1850 ast_sip_destroy_distributor();
1851 ast_res_pjsip_destroy_configuration();
1852 ast_sip_destroy_global_headers();
1853 if (monitor_thread) {
1854 stop_monitor_thread();
1857 pj_pool_release(memory_pool);
1860 if (ast_pjsip_endpoint) {
1861 pjsip_endpt_destroy(ast_pjsip_endpoint);
1862 ast_pjsip_endpoint = NULL;
1864 pj_caching_pool_destroy(&caching_pool);
1865 return AST_MODULE_LOAD_DECLINE;
1868 static int reload_module(void)
1870 if (ast_res_pjsip_reload_configuration()) {
1871 return AST_MODULE_LOAD_DECLINE;
1873 ast_res_pjsip_init_options_handling(1);
1877 static int unload_pjsip(void *data)
1880 pj_pool_release(memory_pool);
1883 if (ast_pjsip_endpoint) {
1884 pjsip_endpt_destroy(ast_pjsip_endpoint);
1885 ast_pjsip_endpoint = NULL;
1887 pj_caching_pool_destroy(&caching_pool);
1891 static int unload_module(void)
1893 ast_res_pjsip_cleanup_options_handling();
1894 ast_sip_destroy_distributor();
1895 ast_res_pjsip_destroy_configuration();
1896 ast_sip_destroy_global_headers();
1897 if (monitor_thread) {
1898 stop_monitor_thread();
1900 /* The thread this is called from cannot call PJSIP/PJLIB functions,
1901 * so we have to push the work to the threadpool to handle
1903 ast_sip_push_task_synchronous(NULL, unload_pjsip, NULL);
1905 ast_threadpool_shutdown(sip_threadpool);
1910 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Basic SIP resource",
1911 .load = load_module,
1912 .unload = unload_module,
1913 .reload = reload_module,
1914 .load_pri = AST_MODPRI_CHANNEL_DEPEND - 5,
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