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>
938 <configOption name="threadpool_initial_size" default="0">
939 <synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
941 <configOption name="threadpool_auto_increment" default="5">
942 <synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
944 <configOption name="threadpool_idle_timeout" default="60">
945 <synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
947 <configOption name="threadpool_max_size" default="0">
948 <synopsis>Maximum number of threads in the res_pjsip threadpool.
949 A value of 0 indicates no maximum.</synopsis>
952 <configObject name="global">
953 <synopsis>Options that apply globally to all SIP communications</synopsis>
955 The settings in this section are global. Unlike options in the <literal>system</literal>
956 section, these options can be refreshed by performing a reload.
957 </para></description>
958 <configOption name="maxforwards" default="70">
959 <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
961 <configOption name="useragent" default="Asterisk <Asterisk Version>">
962 <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
967 <manager name="PJSIPQualify" language="en_US">
969 Qualify a chan_pjsip endpoint.
972 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
973 <parameter name="Endpoint" required="true">
974 <para>The endpoint you want to qualify.</para>
978 <para>Qualify a chan_pjsip endpoint.</para>
984 static pjsip_endpoint *ast_pjsip_endpoint;
986 static struct ast_threadpool *sip_threadpool;
988 static int register_service(void *data)
990 pjsip_module **module = data;
991 if (!ast_pjsip_endpoint) {
992 ast_log(LOG_ERROR, "There is no PJSIP endpoint. Unable to register services\n");
995 if (pjsip_endpt_register_module(ast_pjsip_endpoint, *module) != PJ_SUCCESS) {
996 ast_log(LOG_ERROR, "Unable to register module %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
999 ast_debug(1, "Registered SIP service %.*s (%p)\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name), *module);
1000 ast_module_ref(ast_module_info->self);
1004 int ast_sip_register_service(pjsip_module *module)
1006 return ast_sip_push_task_synchronous(NULL, register_service, &module);
1009 static int unregister_service(void *data)
1011 pjsip_module **module = data;
1012 ast_module_unref(ast_module_info->self);
1013 if (!ast_pjsip_endpoint) {
1016 pjsip_endpt_unregister_module(ast_pjsip_endpoint, *module);
1017 ast_debug(1, "Unregistered SIP service %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1021 void ast_sip_unregister_service(pjsip_module *module)
1023 ast_sip_push_task_synchronous(NULL, unregister_service, &module);
1026 static struct ast_sip_authenticator *registered_authenticator;
1028 int ast_sip_register_authenticator(struct ast_sip_authenticator *auth)
1030 if (registered_authenticator) {
1031 ast_log(LOG_WARNING, "Authenticator %p is already registered. Cannot register a new one\n", registered_authenticator);
1034 registered_authenticator = auth;
1035 ast_debug(1, "Registered SIP authenticator module %p\n", auth);
1036 ast_module_ref(ast_module_info->self);
1040 void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth)
1042 if (registered_authenticator != auth) {
1043 ast_log(LOG_WARNING, "Trying to unregister authenticator %p but authenticator %p registered\n",
1044 auth, registered_authenticator);
1047 registered_authenticator = NULL;
1048 ast_debug(1, "Unregistered SIP authenticator %p\n", auth);
1049 ast_module_unref(ast_module_info->self);
1052 int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1054 if (!registered_authenticator) {
1055 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is not required\n");
1059 return registered_authenticator->requires_authentication(endpoint, rdata);
1062 enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
1063 pjsip_rx_data *rdata, pjsip_tx_data *tdata)
1065 if (!registered_authenticator) {
1066 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is successful\n");
1069 return registered_authenticator->check_authentication(endpoint, rdata, tdata);
1072 static struct ast_sip_outbound_authenticator *registered_outbound_authenticator;
1074 int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1076 if (registered_outbound_authenticator) {
1077 ast_log(LOG_WARNING, "Outbound authenticator %p is already registered. Cannot register a new one\n", registered_outbound_authenticator);
1080 registered_outbound_authenticator = auth;
1081 ast_debug(1, "Registered SIP outbound authenticator module %p\n", auth);
1082 ast_module_ref(ast_module_info->self);
1086 void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1088 if (registered_outbound_authenticator != auth) {
1089 ast_log(LOG_WARNING, "Trying to unregister outbound authenticator %p but outbound authenticator %p registered\n",
1090 auth, registered_outbound_authenticator);
1093 registered_outbound_authenticator = NULL;
1094 ast_debug(1, "Unregistered SIP outbound authenticator %p\n", auth);
1095 ast_module_unref(ast_module_info->self);
1098 int ast_sip_create_request_with_auth(const struct ast_sip_auth_array *auths, pjsip_rx_data *challenge,
1099 pjsip_transaction *tsx, pjsip_tx_data **new_request)
1101 if (!registered_outbound_authenticator) {
1102 ast_log(LOG_WARNING, "No SIP outbound authenticator registered. Cannot respond to authentication challenge\n");
1105 return registered_outbound_authenticator->create_request_with_auth(auths, challenge, tsx, new_request);
1108 struct endpoint_identifier_list {
1109 struct ast_sip_endpoint_identifier *identifier;
1110 AST_RWLIST_ENTRY(endpoint_identifier_list) list;
1113 static AST_RWLIST_HEAD_STATIC(endpoint_identifiers, endpoint_identifier_list);
1115 int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1117 struct endpoint_identifier_list *id_list_item;
1118 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1120 id_list_item = ast_calloc(1, sizeof(*id_list_item));
1121 if (!id_list_item) {
1122 ast_log(LOG_ERROR, "Unabled to add endpoint identifier. Out of memory.\n");
1125 id_list_item->identifier = identifier;
1127 AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list);
1128 ast_debug(1, "Registered endpoint identifier %p\n", identifier);
1130 ast_module_ref(ast_module_info->self);
1134 void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1136 struct endpoint_identifier_list *iter;
1137 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1138 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) {
1139 if (iter->identifier == identifier) {
1140 AST_RWLIST_REMOVE_CURRENT(list);
1142 ast_debug(1, "Unregistered endpoint identifier %p\n", identifier);
1143 ast_module_unref(ast_module_info->self);
1147 AST_RWLIST_TRAVERSE_SAFE_END;
1150 struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata)
1152 struct endpoint_identifier_list *iter;
1153 struct ast_sip_endpoint *endpoint = NULL;
1154 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1155 AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) {
1156 ast_assert(iter->identifier->identify_endpoint != NULL);
1157 endpoint = iter->identifier->identify_endpoint(rdata);
1165 pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
1167 return ast_pjsip_endpoint;
1170 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)
1172 pj_str_t tmp, local_addr;
1174 pjsip_sip_uri *sip_uri;
1175 pjsip_transport_type_e type = PJSIP_TRANSPORT_UNSPECIFIED;
1177 char uuid_str[AST_UUID_STR_LEN];
1179 if (ast_strlen_zero(user)) {
1180 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1184 user = ast_uuid_to_str(uuid, uuid_str, sizeof(uuid_str));
1187 /* Parse the provided target URI so we can determine what transport it will end up using */
1188 pj_strdup_with_null(pool, &tmp, target);
1190 if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) ||
1191 (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
1195 sip_uri = pjsip_uri_get_uri(uri);
1197 /* Determine the transport type to use */
1198 if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
1199 type = PJSIP_TRANSPORT_TLS;
1200 } else if (!sip_uri->transport_param.slen) {
1201 type = PJSIP_TRANSPORT_UDP;
1203 type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
1206 if (type == PJSIP_TRANSPORT_UNSPECIFIED) {
1210 /* If the host is IPv6 turn the transport into an IPv6 version */
1211 if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1212 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1215 if (!ast_strlen_zero(domain)) {
1216 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1217 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1219 (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1222 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1223 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1227 /* Get the local bound address for the transport that will be used when communicating with the provided URI */
1228 if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), pool, type, selector,
1229 &local_addr, &local_port) != PJ_SUCCESS) {
1233 /* If IPv6 was specified in the transport, set the proper type */
1234 if (pj_strchr(&local_addr, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1235 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1238 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1239 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1240 "<%s:%s@%s%.*s%s:%d%s%s>",
1241 (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1243 (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1244 (int)local_addr.slen,
1246 (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1248 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1249 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1254 static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
1256 RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
1257 const char *transport_name = endpoint->transport;
1259 if (ast_strlen_zero(transport_name)) {
1263 transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name);
1265 if (!transport || !transport->state) {
1269 if (transport->state->transport) {
1270 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1271 selector->u.transport = transport->state->transport;
1272 } else if (transport->state->factory) {
1273 selector->type = PJSIP_TPSELECTOR_LISTENER;
1274 selector->u.listener = transport->state->factory;
1282 static int sip_get_tpselector_from_uri(const char *uri, pjsip_tpselector *selector)
1284 RAII_VAR(struct ast_sip_contact_transport *, contact_transport, NULL, ao2_cleanup);
1286 contact_transport = ast_sip_location_retrieve_contact_transport_by_uri(uri);
1288 if (!contact_transport) {
1292 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1293 selector->u.transport = contact_transport->transport;
1298 pjsip_dialog *ast_sip_create_dialog(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
1300 pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri;
1301 pjsip_dialog *dlg = NULL;
1302 const char *outbound_proxy = endpoint->outbound_proxy;
1303 pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1304 static const pj_str_t HCONTACT = { "Contact", 7 };
1306 pj_cstr(&remote_uri, uri);
1308 if (pjsip_dlg_create_uac(pjsip_ua_instance(), &local_uri, NULL, &remote_uri, NULL, &dlg) != PJ_SUCCESS) {
1312 if (sip_get_tpselector_from_uri(uri, &selector) && sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1313 pjsip_dlg_terminate(dlg);
1317 if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
1318 pjsip_dlg_terminate(dlg);
1322 /* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
1323 pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
1324 dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
1325 dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
1327 /* If a request user has been specified and we are permitted to change it, do so */
1328 if (!ast_strlen_zero(request_user) && (PJSIP_URI_SCHEME_IS_SIP(dlg->target) || PJSIP_URI_SCHEME_IS_SIPS(dlg->target))) {
1329 pjsip_sip_uri *target = pjsip_uri_get_uri(dlg->target);
1330 pj_strdup2(dlg->pool, &target->user, request_user);
1333 /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
1336 pjsip_dlg_set_transport(dlg, &selector);
1338 if (!ast_strlen_zero(outbound_proxy)) {
1339 pjsip_route_hdr route_set, *route;
1340 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1343 pj_list_init(&route_set);
1345 pj_strdup2_with_null(dlg->pool, &tmp, outbound_proxy);
1346 if (!(route = pjsip_parse_hdr(dlg->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1347 pjsip_dlg_terminate(dlg);
1350 pj_list_push_back(&route_set, route);
1352 pjsip_dlg_set_route_set(dlg, &route_set);
1360 /* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
1361 const pjsip_method pjsip_info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
1362 const pjsip_method pjsip_message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
1366 const pjsip_method *pmethod;
1368 { "INVITE", &pjsip_invite_method },
1369 { "CANCEL", &pjsip_cancel_method },
1370 { "ACK", &pjsip_ack_method },
1371 { "BYE", &pjsip_bye_method },
1372 { "REGISTER", &pjsip_register_method },
1373 { "OPTIONS", &pjsip_options_method },
1374 { "SUBSCRIBE", &pjsip_subscribe_method },
1375 { "NOTIFY", &pjsip_notify_method },
1376 { "PUBLISH", &pjsip_publish_method },
1377 { "INFO", &pjsip_info_method },
1378 { "MESSAGE", &pjsip_message_method },
1381 static const pjsip_method *get_pjsip_method(const char *method)
1384 for (i = 0; i < ARRAY_LEN(methods); ++i) {
1385 if (!strcmp(method, methods[i].method)) {
1386 return methods[i].pmethod;
1392 static int create_in_dialog_request(const pjsip_method *method, struct pjsip_dialog *dlg, pjsip_tx_data **tdata)
1394 if (pjsip_dlg_create_request(dlg, method, -1, tdata) != PJ_SUCCESS) {
1395 ast_log(LOG_WARNING, "Unable to create in-dialog request.\n");
1402 static int create_out_of_dialog_request(const pjsip_method *method, struct ast_sip_endpoint *endpoint,
1403 const char *uri, pjsip_tx_data **tdata)
1405 RAII_VAR(struct ast_sip_contact *, contact, NULL, ao2_cleanup);
1406 pj_str_t remote_uri;
1409 pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1411 if (ast_strlen_zero(uri)) {
1413 ast_log(LOG_ERROR, "An endpoint and/or uri must be specified\n");
1417 contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
1418 if (!contact || ast_strlen_zero(contact->uri)) {
1419 ast_log(LOG_ERROR, "Unable to retrieve contact for endpoint %s\n",
1420 ast_sorcery_object_get_id(endpoint));
1424 pj_cstr(&remote_uri, contact->uri);
1426 pj_cstr(&remote_uri, uri);
1430 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1431 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
1432 ast_sorcery_object_get_id(endpoint));
1437 pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
1440 ast_log(LOG_ERROR, "Unable to create PJLIB memory pool\n");
1444 if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL,
1445 endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
1446 ast_log(LOG_ERROR, "Unable to create From header for %.*s request to endpoint %s\n",
1447 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1448 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1452 if (pjsip_endpt_create_request(ast_sip_get_pjsip_endpoint(), method, &remote_uri,
1453 &from, &remote_uri, &from, NULL, -1, NULL, tdata) != PJ_SUCCESS) {
1454 ast_log(LOG_ERROR, "Unable to create outbound %.*s request to endpoint %s\n",
1455 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1456 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1460 /* We can release this pool since request creation copied all the necessary
1461 * data into the outbound request's pool
1463 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1467 int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
1468 struct ast_sip_endpoint *endpoint, const char *uri,
1469 pjsip_tx_data **tdata)
1471 const pjsip_method *pmethod = get_pjsip_method(method);
1474 ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
1479 return create_in_dialog_request(pmethod, dlg, tdata);
1481 return create_out_of_dialog_request(pmethod, endpoint, uri, tdata);
1485 static int send_in_dialog_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg)
1487 if (pjsip_dlg_send_request(dlg, tdata, -1, NULL) != PJ_SUCCESS) {
1488 ast_log(LOG_WARNING, "Unable to send in-dialog request.\n");
1494 static void send_request_cb(void *token, pjsip_event *e)
1496 RAII_VAR(struct ast_sip_endpoint *, endpoint, token, ao2_cleanup);
1497 pjsip_transaction *tsx = e->body.tsx_state.tsx;
1498 pjsip_rx_data *challenge = e->body.tsx_state.src.rdata;
1499 pjsip_tx_data *tdata;
1501 if (tsx->status_code != 401 && tsx->status_code != 407) {
1505 if (!ast_sip_create_request_with_auth(&endpoint->outbound_auths, challenge, tsx, &tdata)) {
1506 pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, NULL, NULL);
1510 static int send_out_of_dialog_request(pjsip_tx_data *tdata, struct ast_sip_endpoint *endpoint)
1512 ao2_ref(endpoint, +1);
1513 if (pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, endpoint, send_request_cb) != PJ_SUCCESS) {
1514 ast_log(LOG_ERROR, "Error attempting to send outbound %.*s request to endpoint %s\n",
1515 (int) pj_strlen(&tdata->msg->line.req.method.name),
1516 pj_strbuf(&tdata->msg->line.req.method.name),
1517 ast_sorcery_object_get_id(endpoint));
1518 ao2_ref(endpoint, -1);
1525 int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint)
1527 ast_assert(tdata->msg->type == PJSIP_REQUEST_MSG);
1530 return send_in_dialog_request(tdata, dlg);
1532 return send_out_of_dialog_request(tdata, endpoint);
1536 int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value)
1540 pjsip_generic_string_hdr *hdr;
1542 pj_cstr(&hdr_name, name);
1543 pj_cstr(&hdr_value, value);
1545 hdr = pjsip_generic_string_hdr_create(tdata->pool, &hdr_name, &hdr_value);
1547 pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) hdr);
1551 static pjsip_msg_body *ast_body_to_pjsip_body(pj_pool_t *pool, const struct ast_sip_body *body)
1557 pj_cstr(&type, body->type);
1558 pj_cstr(&subtype, body->subtype);
1559 pj_cstr(&body_text, body->body_text);
1561 return pjsip_msg_body_create(pool, &type, &subtype, &body_text);
1564 int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body)
1566 pjsip_msg_body *pjsip_body = ast_body_to_pjsip_body(tdata->pool, body);
1567 tdata->msg->body = pjsip_body;
1571 int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies)
1574 /* NULL for type and subtype automatically creates "multipart/mixed" */
1575 pjsip_msg_body *body = pjsip_multipart_create(tdata->pool, NULL, NULL);
1577 for (i = 0; i < num_bodies; ++i) {
1578 pjsip_multipart_part *part = pjsip_multipart_create_part(tdata->pool);
1579 part->body = ast_body_to_pjsip_body(tdata->pool, bodies[i]);
1580 pjsip_multipart_add_part(tdata->pool, body, part);
1583 tdata->msg->body = body;
1587 int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text)
1589 size_t combined_size = strlen(body_text) + tdata->msg->body->len;
1590 struct ast_str *body_buffer = ast_str_alloca(combined_size);
1592 ast_str_set(&body_buffer, 0, "%.*s%s", (int) tdata->msg->body->len, (char *) tdata->msg->body->data, body_text);
1594 tdata->msg->body->data = pj_pool_alloc(tdata->pool, combined_size);
1595 pj_memcpy(tdata->msg->body->data, ast_str_buffer(body_buffer), combined_size);
1596 tdata->msg->body->len = combined_size;
1601 struct ast_taskprocessor *ast_sip_create_serializer(void)
1603 struct ast_taskprocessor *serializer;
1604 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1605 char name[AST_UUID_STR_LEN];
1611 ast_uuid_to_str(uuid, name, sizeof(name));
1613 serializer = ast_threadpool_serializer(name, sip_threadpool);
1620 int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1623 return ast_taskprocessor_push(serializer, sip_task, task_data);
1625 return ast_threadpool_push(sip_threadpool, sip_task, task_data);
1629 struct sync_task_data {
1634 int (*task)(void *);
1638 static int sync_task(void *data)
1640 struct sync_task_data *std = data;
1641 std->fail = std->task(std->task_data);
1643 ast_mutex_lock(&std->lock);
1645 ast_cond_signal(&std->cond);
1646 ast_mutex_unlock(&std->lock);
1650 int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1652 /* This method is an onion */
1653 struct sync_task_data std;
1654 ast_mutex_init(&std.lock);
1655 ast_cond_init(&std.cond, NULL);
1656 std.fail = std.complete = 0;
1657 std.task = sip_task;
1658 std.task_data = task_data;
1661 if (ast_taskprocessor_push(serializer, sync_task, &std)) {
1665 if (ast_threadpool_push(sip_threadpool, sync_task, &std)) {
1670 ast_mutex_lock(&std.lock);
1671 while (!std.complete) {
1672 ast_cond_wait(&std.cond, &std.lock);
1674 ast_mutex_unlock(&std.lock);
1676 ast_mutex_destroy(&std.lock);
1677 ast_cond_destroy(&std.cond);
1681 void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
1683 size_t chars_to_copy = MIN(size - 1, pj_strlen(src));
1684 memcpy(dest, pj_strbuf(src), chars_to_copy);
1685 dest[chars_to_copy] = '\0';
1688 int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype)
1690 pjsip_media_type compare;
1692 if (!content_type) {
1696 pjsip_media_type_init2(&compare, type, subtype);
1698 return pjsip_media_type_cmp(content_type, &compare, 0) ? -1 : 0;
1701 pj_caching_pool caching_pool;
1702 pj_pool_t *memory_pool;
1703 pj_thread_t *monitor_thread;
1704 static int monitor_continue;
1706 static void *monitor_thread_exec(void *endpt)
1708 while (monitor_continue) {
1709 const pj_time_val delay = {0, 10};
1710 pjsip_endpt_handle_events(ast_pjsip_endpoint, &delay);
1715 static void stop_monitor_thread(void)
1717 monitor_continue = 0;
1718 pj_thread_join(monitor_thread);
1721 AST_THREADSTORAGE(pj_thread_storage);
1722 AST_THREADSTORAGE(servant_id_storage);
1723 #define SIP_SERVANT_ID 0x5E2F1D
1725 static void sip_thread_start(void)
1727 pj_thread_desc *desc;
1728 pj_thread_t *thread;
1729 uint32_t *servant_id;
1731 servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1733 ast_log(LOG_ERROR, "Could not set SIP servant ID in thread-local storage.\n");
1736 *servant_id = SIP_SERVANT_ID;
1738 desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc));
1740 ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n");
1743 pj_bzero(*desc, sizeof(*desc));
1745 if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) {
1746 ast_log(LOG_ERROR, "Couldn't register thread with PJLIB.\n");
1750 int ast_sip_thread_is_servant(void)
1752 uint32_t *servant_id;
1754 servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1759 return *servant_id == SIP_SERVANT_ID;
1762 static void remove_request_headers(pjsip_endpoint *endpt)
1764 const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt);
1765 pjsip_hdr *iter = request_headers->next;
1767 while (iter != request_headers) {
1768 pjsip_hdr *to_erase = iter;
1770 pj_list_erase(to_erase);
1774 static int load_module(void)
1776 /* The third parameter is just copied from
1777 * example code from PJLIB. This can be adjusted
1781 struct ast_threadpool_options options;
1783 if (pj_init() != PJ_SUCCESS) {
1784 return AST_MODULE_LOAD_DECLINE;
1787 if (pjlib_util_init() != PJ_SUCCESS) {
1789 return AST_MODULE_LOAD_DECLINE;
1792 pj_caching_pool_init(&caching_pool, NULL, 1024 * 1024);
1793 if (pjsip_endpt_create(&caching_pool.factory, "SIP", &ast_pjsip_endpoint) != PJ_SUCCESS) {
1794 ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n");
1798 /* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that,
1799 * we need to stop PJSIP from doing it automatically
1801 remove_request_headers(ast_pjsip_endpoint);
1803 memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL);
1805 ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n");
1809 if (ast_sip_initialize_system()) {
1810 ast_log(LOG_ERROR, "Failed to initialize SIP system configuration. Aborting load\n");
1814 sip_get_threadpool_options(&options);
1815 options.thread_start = sip_thread_start;
1816 sip_threadpool = ast_threadpool_create("SIP", NULL, &options);
1817 if (!sip_threadpool) {
1818 ast_log(LOG_ERROR, "Failed to create SIP threadpool. Aborting load\n");
1822 pjsip_tsx_layer_init_module(ast_pjsip_endpoint);
1823 pjsip_ua_init_module(ast_pjsip_endpoint, NULL);
1825 monitor_continue = 1;
1826 status = pj_thread_create(memory_pool, "SIP", (pj_thread_proc *) &monitor_thread_exec,
1827 NULL, PJ_THREAD_DEFAULT_STACK_SIZE * 2, 0, &monitor_thread);
1828 if (status != PJ_SUCCESS) {
1829 ast_log(LOG_ERROR, "Failed to start SIP monitor thread. Aborting load\n");
1833 ast_sip_initialize_global_headers();
1835 if (ast_res_pjsip_initialize_configuration()) {
1836 ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
1840 if (ast_sip_initialize_distributor()) {
1841 ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n");
1845 if (ast_sip_initialize_outbound_authentication()) {
1846 ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n");
1850 ast_res_pjsip_init_options_handling(0);
1852 ast_res_pjsip_init_contact_transports();
1854 return AST_MODULE_LOAD_SUCCESS;
1857 ast_sip_destroy_distributor();
1858 ast_res_pjsip_destroy_configuration();
1859 ast_sip_destroy_global_headers();
1860 if (monitor_thread) {
1861 stop_monitor_thread();
1864 pj_pool_release(memory_pool);
1867 if (ast_pjsip_endpoint) {
1868 pjsip_endpt_destroy(ast_pjsip_endpoint);
1869 ast_pjsip_endpoint = NULL;
1871 pj_caching_pool_destroy(&caching_pool);
1872 return AST_MODULE_LOAD_DECLINE;
1875 static int reload_module(void)
1877 if (ast_res_pjsip_reload_configuration()) {
1878 return AST_MODULE_LOAD_DECLINE;
1880 ast_res_pjsip_init_options_handling(1);
1884 static int unload_pjsip(void *data)
1887 pj_pool_release(memory_pool);
1890 if (ast_pjsip_endpoint) {
1891 pjsip_endpt_destroy(ast_pjsip_endpoint);
1892 ast_pjsip_endpoint = NULL;
1894 pj_caching_pool_destroy(&caching_pool);
1898 static int unload_module(void)
1900 ast_res_pjsip_cleanup_options_handling();
1901 ast_sip_destroy_distributor();
1902 ast_res_pjsip_destroy_configuration();
1903 ast_sip_destroy_global_headers();
1904 if (monitor_thread) {
1905 stop_monitor_thread();
1907 /* The thread this is called from cannot call PJSIP/PJLIB functions,
1908 * so we have to push the work to the threadpool to handle
1910 ast_sip_push_task_synchronous(NULL, unload_pjsip, NULL);
1912 ast_threadpool_shutdown(sip_threadpool);
1917 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Basic SIP resource",
1918 .load = load_module,
1919 .unload = unload_module,
1920 .reload = reload_module,
1921 .load_pri = AST_MODPRI_CHANNEL_DEPEND - 5,
projects / asterisk/asterisk.git / blob