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 authentication information for use
627 by other objects such as <literal>endpoints</literal> or <literal>registrations</literal>.
628 This also allows for multiple objects to use a single auth object. See
629 the <literal>auth_type</literal> config option for password style choices.
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 when trying to authenticate an endpoint inbound request. If set to <literal>userpass</literal>
636 then we'll read from the 'password' option. For <literal>md5</literal> we'll read
641 <enum name="userpass"/>
645 <configOption name="nonce_lifetime" default="32">
646 <synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
648 <configOption name="md5_cred">
649 <synopsis>MD5 Hash used for authentication.</synopsis>
650 <description><para>Only used when auth_type is <literal>md5</literal>.</para></description>
652 <configOption name="password">
653 <synopsis>PlainText password used for authentication.</synopsis>
654 <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
656 <configOption name="realm" default="asterisk">
657 <synopsis>SIP realm for endpoint</synopsis>
659 <configOption name="type">
660 <synopsis>Must be 'auth'</synopsis>
662 <configOption name="username">
663 <synopsis>Username to use for account</synopsis>
666 <configObject name="nat_hook">
667 <synopsis>XXX This exists only to prevent XML documentation errors.</synopsis>
668 <configOption name="external_media_address">
669 <synopsis>I should be undocumented or hidden</synopsis>
671 <configOption name="method">
672 <synopsis>I should be undocumented or hidden</synopsis>
675 <configObject name="domain_alias">
676 <synopsis>Domain Alias</synopsis>
678 Signifies that a domain is an alias. If the domain on a session is
679 not found to match an AoR then this object is used to see if we have
680 an alias for the AoR to which the endpoint is binding. This objects
681 name as defined in configuration should be the domain alias and a
682 config option is provided to specify the domain to be aliased.
683 </para></description>
684 <configOption name="type">
685 <synopsis>Must be of type 'domain_alias'.</synopsis>
687 <configOption name="domain">
688 <synopsis>Domain to be aliased</synopsis>
691 <configObject name="transport">
692 <synopsis>SIP Transport</synopsis>
694 <emphasis>Transports</emphasis>
696 <para>There are different transports and protocol derivatives
697 supported by <literal>res_pjsip</literal>. They are in order of
698 preference: UDP, TCP, and WebSocket (WS).</para>
699 <note><para>Changes to transport configuration in pjsip.conf will only be
700 effected on a complete restart of Asterisk. A module reload
701 will not suffice.</para></note>
703 <configOption name="async_operations" default="1">
704 <synopsis>Number of simultaneous Asynchronous Operations</synopsis>
706 <configOption name="bind">
707 <synopsis>IP Address and optional port to bind to for this transport</synopsis>
709 <configOption name="ca_list_file">
710 <synopsis>File containing a list of certificates to read (TLS ONLY)</synopsis>
712 <configOption name="cert_file">
713 <synopsis>Certificate file for endpoint (TLS ONLY)</synopsis>
715 <configOption name="cipher">
716 <synopsis>Preferred Cryptography Cipher (TLS ONLY)</synopsis>
718 Many options for acceptable ciphers see link for more:
719 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
720 </para></description>
722 <configOption name="domain">
723 <synopsis>Domain the transport comes from</synopsis>
725 <configOption name="external_media_address">
726 <synopsis>External Address to use in RTP handling</synopsis>
728 <configOption name="external_signaling_address">
729 <synopsis>External address for SIP signalling</synopsis>
731 <configOption name="external_signaling_port" default="0">
732 <synopsis>External port for SIP signalling</synopsis>
734 <configOption name="method">
735 <synopsis>Method of SSL transport (TLS ONLY)</synopsis>
738 <enum name="default" />
739 <enum name="unspecified" />
740 <enum name="tlsv1" />
741 <enum name="sslv2" />
742 <enum name="sslv3" />
743 <enum name="sslv23" />
747 <configOption name="localnet">
748 <synopsis>Network to consider local (used for NAT purposes).</synopsis>
749 <description><para>This must be in CIDR or dotted decimal format with the IP
750 and mask separated with a slash ('/').</para></description>
752 <configOption name="password">
753 <synopsis>Password required for transport</synopsis>
755 <configOption name="privkey_file">
756 <synopsis>Private key file (TLS ONLY)</synopsis>
758 <configOption name="protocol" default="udp">
759 <synopsis>Protocol to use for SIP traffic</synopsis>
768 <configOption name="require_client_cert" default="false">
769 <synopsis>Require client certificate (TLS ONLY)</synopsis>
771 <configOption name="type">
772 <synopsis>Must be of type 'transport'.</synopsis>
774 <configOption name="verify_client" default="false">
775 <synopsis>Require verification of client certificate (TLS ONLY)</synopsis>
777 <configOption name="verify_server" default="false">
778 <synopsis>Require verification of server certificate (TLS ONLY)</synopsis>
781 <configObject name="contact">
782 <synopsis>A way of creating an aliased name to a SIP URI</synopsis>
784 Contacts are a way to hide SIP URIs from the dialplan directly.
785 They are also used to make a group of contactable parties when
786 in use with <literal>AoR</literal> lists.
787 </para></description>
788 <configOption name="type">
789 <synopsis>Must be of type 'contact'.</synopsis>
791 <configOption name="uri">
792 <synopsis>SIP URI to contact peer</synopsis>
794 <configOption name="expiration_time">
795 <synopsis>Time to keep alive a contact</synopsis>
797 Time to keep alive a contact. String style specification.
798 </para></description>
800 <configOption name="qualify_frequency" default="0">
801 <synopsis>Interval at which to qualify a contact</synopsis>
803 Interval between attempts to qualify the contact for reachability.
804 If <literal>0</literal> never qualify. Time in seconds.
805 </para></description>
808 <configObject name="contact_status">
809 <synopsis>Status for a contact</synopsis>
811 The contact status keeps track of whether or not a contact is reachable
812 and how long it took to qualify the contact (round trip time).
813 </para></description>
814 <configOption name="status">
815 <synopsis>A contact's status</synopsis>
818 <enum name="AVAILABLE" />
819 <enum name="UNAVAILABLE" />
823 <configOption name="rtt">
824 <synopsis>Round trip time</synopsis>
826 The time, in microseconds, it took to qualify the contact.
827 </para></description>
830 <configObject name="aor">
831 <synopsis>The configuration for a location of an endpoint</synopsis>
833 An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
834 AoRs are specified, an endpoint will not be reachable by Asterisk.
835 Beyond that, an AoR has other uses within Asterisk, such as inbound
838 An <literal>AoR</literal> is a way to allow dialing a group
839 of <literal>Contacts</literal> that all use the same
840 <literal>endpoint</literal> for calls.
842 This can be used as another way of grouping a list of contacts to dial
843 rather than specifing them each directly when dialing via the dialplan.
844 This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
846 Registrations: For Asterisk to match an inbound registration to an endpoint,
847 the AoR object name must match the user portion of the SIP URI in the "To:"
848 header of the inbound SIP registration. That will usually be equivalent
849 to the "user name" set in your hard or soft phones configuration.
850 </para></description>
851 <configOption name="contact">
852 <synopsis>Permanent contacts assigned to AoR</synopsis>
854 Contacts specified will be called whenever referenced
855 by <literal>chan_pjsip</literal>.
857 Use a separate "contact=" entry for each contact required. Contacts
858 are specified using a SIP URI.
859 </para></description>
861 <configOption name="default_expiration" default="3600">
862 <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
864 <configOption name="mailboxes">
865 <synopsis>Mailbox(es) to be associated with</synopsis>
866 <description><para>This option applies when an external entity subscribes to an AoR
867 for message waiting indications. The mailboxes specified will be subscribed to.
868 More than one mailbox can be specified with a comma-delimited string.</para></description>
870 <configOption name="maximum_expiration" default="7200">
871 <synopsis>Maximum time to keep an AoR</synopsis>
873 Maximium time to keep a peer with explicit expiration. Time in seconds.
874 </para></description>
876 <configOption name="max_contacts" default="0">
877 <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
879 Maximum number of contacts that can associate with this AoR. This value does
880 not affect the number of contacts that can be added with the "contact" option.
881 It only limits contacts added through external interaction, such as
884 <note><para>This should be set to <literal>1</literal> and
885 <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
886 wish to stick with the older <literal>chan_sip</literal> behaviour.
890 <configOption name="minimum_expiration" default="60">
891 <synopsis>Minimum keep alive time for an AoR</synopsis>
893 Minimum time to keep a peer with an explict expiration. Time in seconds.
894 </para></description>
896 <configOption name="remove_existing" default="no">
897 <synopsis>Determines whether new contacts replace existing ones.</synopsis>
899 On receiving a new registration to the AoR should it remove
900 the existing contact that was registered against it?
902 <note><para>This should be set to <literal>yes</literal> and
903 <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
904 wish to stick with the older <literal>chan_sip</literal> behaviour.
908 <configOption name="type">
909 <synopsis>Must be of type 'aor'.</synopsis>
911 <configOption name="qualify_frequency" default="0">
912 <synopsis>Interval at which to qualify an AoR</synopsis>
914 Interval between attempts to qualify the AoR for reachability.
915 If <literal>0</literal> never qualify. Time in seconds.
916 </para></description>
918 <configOption name="authenticate_qualify" default="no">
919 <synopsis>Authenticates a qualify request if needed</synopsis>
921 If true and a qualify request receives a challenge or authenticate response
922 authentication is attempted before declaring the contact available.
923 </para></description>
926 <configObject name="system">
927 <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
929 The settings in this section are global. In addition to being global, the values will
930 not be re-evaluated when a reload is performed. This is because the values must be set
931 before the SIP stack is initialized. The only way to reset these values is to either
932 restart Asterisk, or unload res_pjsip.so and then load it again.
933 </para></description>
934 <configOption name="timert1" default="500">
935 <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
937 Timer T1 is the base for determining how long to wait before retransmitting
938 requests that receive no response when using an unreliable transport (e.g. UDP).
939 For more information on this timer, see RFC 3261, Section 17.1.1.1.
940 </para></description>
942 <configOption name="timerb" default="32000">
943 <synopsis>Set transaction timer B value (milliseconds).</synopsis>
945 Timer B determines the maximum amount of time to wait after sending an INVITE
946 request before terminating the transaction. It is recommended that this be set
947 to 64 * Timer T1, but it may be set higher if desired. For more information on
948 this timer, see RFC 3261, Section 17.1.1.1.
949 </para></description>
951 <configOption name="compactheaders" default="no">
952 <synopsis>Use the short forms of common SIP header names.</synopsis>
954 <configOption name="threadpool_initial_size" default="0">
955 <synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
957 <configOption name="threadpool_auto_increment" default="5">
958 <synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
960 <configOption name="threadpool_idle_timeout" default="60">
961 <synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
963 <configOption name="threadpool_max_size" default="0">
964 <synopsis>Maximum number of threads in the res_pjsip threadpool.
965 A value of 0 indicates no maximum.</synopsis>
968 <configObject name="global">
969 <synopsis>Options that apply globally to all SIP communications</synopsis>
971 The settings in this section are global. Unlike options in the <literal>system</literal>
972 section, these options can be refreshed by performing a reload.
973 </para></description>
974 <configOption name="maxforwards" default="70">
975 <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
977 <configOption name="useragent" default="Asterisk <Asterisk Version>">
978 <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
983 <manager name="PJSIPQualify" language="en_US">
985 Qualify a chan_pjsip endpoint.
988 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
989 <parameter name="Endpoint" required="true">
990 <para>The endpoint you want to qualify.</para>
994 <para>Qualify a chan_pjsip endpoint.</para>
1000 static pjsip_endpoint *ast_pjsip_endpoint;
1002 static struct ast_threadpool *sip_threadpool;
1004 static int register_service(void *data)
1006 pjsip_module **module = data;
1007 if (!ast_pjsip_endpoint) {
1008 ast_log(LOG_ERROR, "There is no PJSIP endpoint. Unable to register services\n");
1011 if (pjsip_endpt_register_module(ast_pjsip_endpoint, *module) != PJ_SUCCESS) {
1012 ast_log(LOG_ERROR, "Unable to register module %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1015 ast_debug(1, "Registered SIP service %.*s (%p)\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name), *module);
1016 ast_module_ref(ast_module_info->self);
1020 int ast_sip_register_service(pjsip_module *module)
1022 return ast_sip_push_task_synchronous(NULL, register_service, &module);
1025 static int unregister_service(void *data)
1027 pjsip_module **module = data;
1028 ast_module_unref(ast_module_info->self);
1029 if (!ast_pjsip_endpoint) {
1032 pjsip_endpt_unregister_module(ast_pjsip_endpoint, *module);
1033 ast_debug(1, "Unregistered SIP service %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1037 void ast_sip_unregister_service(pjsip_module *module)
1039 ast_sip_push_task_synchronous(NULL, unregister_service, &module);
1042 static struct ast_sip_authenticator *registered_authenticator;
1044 int ast_sip_register_authenticator(struct ast_sip_authenticator *auth)
1046 if (registered_authenticator) {
1047 ast_log(LOG_WARNING, "Authenticator %p is already registered. Cannot register a new one\n", registered_authenticator);
1050 registered_authenticator = auth;
1051 ast_debug(1, "Registered SIP authenticator module %p\n", auth);
1052 ast_module_ref(ast_module_info->self);
1056 void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth)
1058 if (registered_authenticator != auth) {
1059 ast_log(LOG_WARNING, "Trying to unregister authenticator %p but authenticator %p registered\n",
1060 auth, registered_authenticator);
1063 registered_authenticator = NULL;
1064 ast_debug(1, "Unregistered SIP authenticator %p\n", auth);
1065 ast_module_unref(ast_module_info->self);
1068 int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1070 if (!registered_authenticator) {
1071 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is not required\n");
1075 return registered_authenticator->requires_authentication(endpoint, rdata);
1078 enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
1079 pjsip_rx_data *rdata, pjsip_tx_data *tdata)
1081 if (!registered_authenticator) {
1082 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is successful\n");
1085 return registered_authenticator->check_authentication(endpoint, rdata, tdata);
1088 static struct ast_sip_outbound_authenticator *registered_outbound_authenticator;
1090 int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1092 if (registered_outbound_authenticator) {
1093 ast_log(LOG_WARNING, "Outbound authenticator %p is already registered. Cannot register a new one\n", registered_outbound_authenticator);
1096 registered_outbound_authenticator = auth;
1097 ast_debug(1, "Registered SIP outbound authenticator module %p\n", auth);
1098 ast_module_ref(ast_module_info->self);
1102 void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1104 if (registered_outbound_authenticator != auth) {
1105 ast_log(LOG_WARNING, "Trying to unregister outbound authenticator %p but outbound authenticator %p registered\n",
1106 auth, registered_outbound_authenticator);
1109 registered_outbound_authenticator = NULL;
1110 ast_debug(1, "Unregistered SIP outbound authenticator %p\n", auth);
1111 ast_module_unref(ast_module_info->self);
1114 int ast_sip_create_request_with_auth(const struct ast_sip_auth_array *auths, pjsip_rx_data *challenge,
1115 pjsip_transaction *tsx, pjsip_tx_data **new_request)
1117 if (!registered_outbound_authenticator) {
1118 ast_log(LOG_WARNING, "No SIP outbound authenticator registered. Cannot respond to authentication challenge\n");
1121 return registered_outbound_authenticator->create_request_with_auth(auths, challenge, tsx, new_request);
1124 struct endpoint_identifier_list {
1125 struct ast_sip_endpoint_identifier *identifier;
1126 AST_RWLIST_ENTRY(endpoint_identifier_list) list;
1129 static AST_RWLIST_HEAD_STATIC(endpoint_identifiers, endpoint_identifier_list);
1131 int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1133 struct endpoint_identifier_list *id_list_item;
1134 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1136 id_list_item = ast_calloc(1, sizeof(*id_list_item));
1137 if (!id_list_item) {
1138 ast_log(LOG_ERROR, "Unabled to add endpoint identifier. Out of memory.\n");
1141 id_list_item->identifier = identifier;
1143 AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list);
1144 ast_debug(1, "Registered endpoint identifier %p\n", identifier);
1146 ast_module_ref(ast_module_info->self);
1150 void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1152 struct endpoint_identifier_list *iter;
1153 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1154 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) {
1155 if (iter->identifier == identifier) {
1156 AST_RWLIST_REMOVE_CURRENT(list);
1158 ast_debug(1, "Unregistered endpoint identifier %p\n", identifier);
1159 ast_module_unref(ast_module_info->self);
1163 AST_RWLIST_TRAVERSE_SAFE_END;
1166 struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata)
1168 struct endpoint_identifier_list *iter;
1169 struct ast_sip_endpoint *endpoint = NULL;
1170 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1171 AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) {
1172 ast_assert(iter->identifier->identify_endpoint != NULL);
1173 endpoint = iter->identifier->identify_endpoint(rdata);
1181 pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
1183 return ast_pjsip_endpoint;
1186 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)
1188 pj_str_t tmp, local_addr;
1190 pjsip_sip_uri *sip_uri;
1191 pjsip_transport_type_e type = PJSIP_TRANSPORT_UNSPECIFIED;
1193 char uuid_str[AST_UUID_STR_LEN];
1195 if (ast_strlen_zero(user)) {
1196 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1200 user = ast_uuid_to_str(uuid, uuid_str, sizeof(uuid_str));
1203 /* Parse the provided target URI so we can determine what transport it will end up using */
1204 pj_strdup_with_null(pool, &tmp, target);
1206 if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) ||
1207 (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
1211 sip_uri = pjsip_uri_get_uri(uri);
1213 /* Determine the transport type to use */
1214 if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
1215 type = PJSIP_TRANSPORT_TLS;
1216 } else if (!sip_uri->transport_param.slen) {
1217 type = PJSIP_TRANSPORT_UDP;
1219 type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
1222 if (type == PJSIP_TRANSPORT_UNSPECIFIED) {
1226 /* If the host is IPv6 turn the transport into an IPv6 version */
1227 if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1228 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1231 if (!ast_strlen_zero(domain)) {
1232 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1233 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1235 (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1238 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1239 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1243 /* Get the local bound address for the transport that will be used when communicating with the provided URI */
1244 if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), pool, type, selector,
1245 &local_addr, &local_port) != PJ_SUCCESS) {
1249 /* If IPv6 was specified in the transport, set the proper type */
1250 if (pj_strchr(&local_addr, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1251 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1254 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1255 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1256 "<%s:%s@%s%.*s%s:%d%s%s>",
1257 (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1259 (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1260 (int)local_addr.slen,
1262 (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1264 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1265 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1270 static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
1272 RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
1273 const char *transport_name = endpoint->transport;
1275 if (ast_strlen_zero(transport_name)) {
1279 transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name);
1281 if (!transport || !transport->state) {
1285 if (transport->state->transport) {
1286 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1287 selector->u.transport = transport->state->transport;
1288 } else if (transport->state->factory) {
1289 selector->type = PJSIP_TPSELECTOR_LISTENER;
1290 selector->u.listener = transport->state->factory;
1298 static int sip_get_tpselector_from_uri(const char *uri, pjsip_tpselector *selector)
1300 RAII_VAR(struct ast_sip_contact_transport *, contact_transport, NULL, ao2_cleanup);
1302 contact_transport = ast_sip_location_retrieve_contact_transport_by_uri(uri);
1304 if (!contact_transport) {
1308 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1309 selector->u.transport = contact_transport->transport;
1314 pjsip_dialog *ast_sip_create_dialog(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
1316 pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri;
1317 pjsip_dialog *dlg = NULL;
1318 const char *outbound_proxy = endpoint->outbound_proxy;
1319 pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1320 static const pj_str_t HCONTACT = { "Contact", 7 };
1322 pj_cstr(&remote_uri, uri);
1324 if (pjsip_dlg_create_uac(pjsip_ua_instance(), &local_uri, NULL, &remote_uri, NULL, &dlg) != PJ_SUCCESS) {
1328 if (sip_get_tpselector_from_uri(uri, &selector) && sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1329 pjsip_dlg_terminate(dlg);
1333 if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
1334 pjsip_dlg_terminate(dlg);
1338 /* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
1339 pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
1340 dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
1341 dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
1343 /* If a request user has been specified and we are permitted to change it, do so */
1344 if (!ast_strlen_zero(request_user) && (PJSIP_URI_SCHEME_IS_SIP(dlg->target) || PJSIP_URI_SCHEME_IS_SIPS(dlg->target))) {
1345 pjsip_sip_uri *target = pjsip_uri_get_uri(dlg->target);
1346 pj_strdup2(dlg->pool, &target->user, request_user);
1349 /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
1352 pjsip_dlg_set_transport(dlg, &selector);
1354 if (!ast_strlen_zero(outbound_proxy)) {
1355 pjsip_route_hdr route_set, *route;
1356 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1359 pj_list_init(&route_set);
1361 pj_strdup2_with_null(dlg->pool, &tmp, outbound_proxy);
1362 if (!(route = pjsip_parse_hdr(dlg->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1363 pjsip_dlg_terminate(dlg);
1366 pj_list_push_back(&route_set, route);
1368 pjsip_dlg_set_route_set(dlg, &route_set);
1376 /* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
1377 const pjsip_method pjsip_info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
1378 const pjsip_method pjsip_message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
1382 const pjsip_method *pmethod;
1384 { "INVITE", &pjsip_invite_method },
1385 { "CANCEL", &pjsip_cancel_method },
1386 { "ACK", &pjsip_ack_method },
1387 { "BYE", &pjsip_bye_method },
1388 { "REGISTER", &pjsip_register_method },
1389 { "OPTIONS", &pjsip_options_method },
1390 { "SUBSCRIBE", &pjsip_subscribe_method },
1391 { "NOTIFY", &pjsip_notify_method },
1392 { "PUBLISH", &pjsip_publish_method },
1393 { "INFO", &pjsip_info_method },
1394 { "MESSAGE", &pjsip_message_method },
1397 static const pjsip_method *get_pjsip_method(const char *method)
1400 for (i = 0; i < ARRAY_LEN(methods); ++i) {
1401 if (!strcmp(method, methods[i].method)) {
1402 return methods[i].pmethod;
1408 static int create_in_dialog_request(const pjsip_method *method, struct pjsip_dialog *dlg, pjsip_tx_data **tdata)
1410 if (pjsip_dlg_create_request(dlg, method, -1, tdata) != PJ_SUCCESS) {
1411 ast_log(LOG_WARNING, "Unable to create in-dialog request.\n");
1418 static int create_out_of_dialog_request(const pjsip_method *method, struct ast_sip_endpoint *endpoint,
1419 const char *uri, pjsip_tx_data **tdata)
1421 RAII_VAR(struct ast_sip_contact *, contact, NULL, ao2_cleanup);
1422 pj_str_t remote_uri;
1425 pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1427 if (ast_strlen_zero(uri)) {
1429 ast_log(LOG_ERROR, "An endpoint and/or uri must be specified\n");
1433 contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
1434 if (!contact || ast_strlen_zero(contact->uri)) {
1435 ast_log(LOG_ERROR, "Unable to retrieve contact for endpoint %s\n",
1436 ast_sorcery_object_get_id(endpoint));
1440 pj_cstr(&remote_uri, contact->uri);
1442 pj_cstr(&remote_uri, uri);
1446 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1447 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
1448 ast_sorcery_object_get_id(endpoint));
1453 pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
1456 ast_log(LOG_ERROR, "Unable to create PJLIB memory pool\n");
1460 if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL,
1461 endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
1462 ast_log(LOG_ERROR, "Unable to create From header for %.*s request to endpoint %s\n",
1463 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1464 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1468 if (pjsip_endpt_create_request(ast_sip_get_pjsip_endpoint(), method, &remote_uri,
1469 &from, &remote_uri, &from, NULL, -1, NULL, tdata) != PJ_SUCCESS) {
1470 ast_log(LOG_ERROR, "Unable to create outbound %.*s request to endpoint %s\n",
1471 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1472 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1476 /* We can release this pool since request creation copied all the necessary
1477 * data into the outbound request's pool
1479 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1483 int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
1484 struct ast_sip_endpoint *endpoint, const char *uri,
1485 pjsip_tx_data **tdata)
1487 const pjsip_method *pmethod = get_pjsip_method(method);
1490 ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
1495 return create_in_dialog_request(pmethod, dlg, tdata);
1497 return create_out_of_dialog_request(pmethod, endpoint, uri, tdata);
1501 static int send_in_dialog_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg)
1503 if (pjsip_dlg_send_request(dlg, tdata, -1, NULL) != PJ_SUCCESS) {
1504 ast_log(LOG_WARNING, "Unable to send in-dialog request.\n");
1510 static void send_request_cb(void *token, pjsip_event *e)
1512 RAII_VAR(struct ast_sip_endpoint *, endpoint, token, ao2_cleanup);
1513 pjsip_transaction *tsx = e->body.tsx_state.tsx;
1514 pjsip_rx_data *challenge = e->body.tsx_state.src.rdata;
1515 pjsip_tx_data *tdata;
1517 if (tsx->status_code != 401 && tsx->status_code != 407) {
1521 if (!ast_sip_create_request_with_auth(&endpoint->outbound_auths, challenge, tsx, &tdata)) {
1522 pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, NULL, NULL);
1526 static int send_out_of_dialog_request(pjsip_tx_data *tdata, struct ast_sip_endpoint *endpoint)
1528 ao2_ref(endpoint, +1);
1529 if (pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, endpoint, send_request_cb) != PJ_SUCCESS) {
1530 ast_log(LOG_ERROR, "Error attempting to send outbound %.*s request to endpoint %s\n",
1531 (int) pj_strlen(&tdata->msg->line.req.method.name),
1532 pj_strbuf(&tdata->msg->line.req.method.name),
1533 ast_sorcery_object_get_id(endpoint));
1534 ao2_ref(endpoint, -1);
1541 int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint)
1543 ast_assert(tdata->msg->type == PJSIP_REQUEST_MSG);
1546 return send_in_dialog_request(tdata, dlg);
1548 return send_out_of_dialog_request(tdata, endpoint);
1552 int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value)
1556 pjsip_generic_string_hdr *hdr;
1558 pj_cstr(&hdr_name, name);
1559 pj_cstr(&hdr_value, value);
1561 hdr = pjsip_generic_string_hdr_create(tdata->pool, &hdr_name, &hdr_value);
1563 pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) hdr);
1567 static pjsip_msg_body *ast_body_to_pjsip_body(pj_pool_t *pool, const struct ast_sip_body *body)
1573 pj_cstr(&type, body->type);
1574 pj_cstr(&subtype, body->subtype);
1575 pj_cstr(&body_text, body->body_text);
1577 return pjsip_msg_body_create(pool, &type, &subtype, &body_text);
1580 int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body)
1582 pjsip_msg_body *pjsip_body = ast_body_to_pjsip_body(tdata->pool, body);
1583 tdata->msg->body = pjsip_body;
1587 int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies)
1590 /* NULL for type and subtype automatically creates "multipart/mixed" */
1591 pjsip_msg_body *body = pjsip_multipart_create(tdata->pool, NULL, NULL);
1593 for (i = 0; i < num_bodies; ++i) {
1594 pjsip_multipart_part *part = pjsip_multipart_create_part(tdata->pool);
1595 part->body = ast_body_to_pjsip_body(tdata->pool, bodies[i]);
1596 pjsip_multipart_add_part(tdata->pool, body, part);
1599 tdata->msg->body = body;
1603 int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text)
1605 size_t combined_size = strlen(body_text) + tdata->msg->body->len;
1606 struct ast_str *body_buffer = ast_str_alloca(combined_size);
1608 ast_str_set(&body_buffer, 0, "%.*s%s", (int) tdata->msg->body->len, (char *) tdata->msg->body->data, body_text);
1610 tdata->msg->body->data = pj_pool_alloc(tdata->pool, combined_size);
1611 pj_memcpy(tdata->msg->body->data, ast_str_buffer(body_buffer), combined_size);
1612 tdata->msg->body->len = combined_size;
1617 struct ast_taskprocessor *ast_sip_create_serializer(void)
1619 struct ast_taskprocessor *serializer;
1620 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1621 char name[AST_UUID_STR_LEN];
1627 ast_uuid_to_str(uuid, name, sizeof(name));
1629 serializer = ast_threadpool_serializer(name, sip_threadpool);
1636 int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1639 return ast_taskprocessor_push(serializer, sip_task, task_data);
1641 return ast_threadpool_push(sip_threadpool, sip_task, task_data);
1645 struct sync_task_data {
1650 int (*task)(void *);
1654 static int sync_task(void *data)
1656 struct sync_task_data *std = data;
1657 std->fail = std->task(std->task_data);
1659 ast_mutex_lock(&std->lock);
1661 ast_cond_signal(&std->cond);
1662 ast_mutex_unlock(&std->lock);
1666 int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1668 /* This method is an onion */
1669 struct sync_task_data std;
1670 ast_mutex_init(&std.lock);
1671 ast_cond_init(&std.cond, NULL);
1672 std.fail = std.complete = 0;
1673 std.task = sip_task;
1674 std.task_data = task_data;
1677 if (ast_taskprocessor_push(serializer, sync_task, &std)) {
1681 if (ast_threadpool_push(sip_threadpool, sync_task, &std)) {
1686 ast_mutex_lock(&std.lock);
1687 while (!std.complete) {
1688 ast_cond_wait(&std.cond, &std.lock);
1690 ast_mutex_unlock(&std.lock);
1692 ast_mutex_destroy(&std.lock);
1693 ast_cond_destroy(&std.cond);
1697 void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
1699 size_t chars_to_copy = MIN(size - 1, pj_strlen(src));
1700 memcpy(dest, pj_strbuf(src), chars_to_copy);
1701 dest[chars_to_copy] = '\0';
1704 int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype)
1706 pjsip_media_type compare;
1708 if (!content_type) {
1712 pjsip_media_type_init2(&compare, type, subtype);
1714 return pjsip_media_type_cmp(content_type, &compare, 0) ? -1 : 0;
1717 pj_caching_pool caching_pool;
1718 pj_pool_t *memory_pool;
1719 pj_thread_t *monitor_thread;
1720 static int monitor_continue;
1722 static void *monitor_thread_exec(void *endpt)
1724 while (monitor_continue) {
1725 const pj_time_val delay = {0, 10};
1726 pjsip_endpt_handle_events(ast_pjsip_endpoint, &delay);
1731 static void stop_monitor_thread(void)
1733 monitor_continue = 0;
1734 pj_thread_join(monitor_thread);
1737 AST_THREADSTORAGE(pj_thread_storage);
1738 AST_THREADSTORAGE(servant_id_storage);
1739 #define SIP_SERVANT_ID 0x5E2F1D
1741 static void sip_thread_start(void)
1743 pj_thread_desc *desc;
1744 pj_thread_t *thread;
1745 uint32_t *servant_id;
1747 servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1749 ast_log(LOG_ERROR, "Could not set SIP servant ID in thread-local storage.\n");
1752 *servant_id = SIP_SERVANT_ID;
1754 desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc));
1756 ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n");
1759 pj_bzero(*desc, sizeof(*desc));
1761 if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) {
1762 ast_log(LOG_ERROR, "Couldn't register thread with PJLIB.\n");
1766 int ast_sip_thread_is_servant(void)
1768 uint32_t *servant_id;
1770 servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1775 return *servant_id == SIP_SERVANT_ID;
1778 static void remove_request_headers(pjsip_endpoint *endpt)
1780 const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt);
1781 pjsip_hdr *iter = request_headers->next;
1783 while (iter != request_headers) {
1784 pjsip_hdr *to_erase = iter;
1786 pj_list_erase(to_erase);
1790 static int load_module(void)
1792 /* The third parameter is just copied from
1793 * example code from PJLIB. This can be adjusted
1797 struct ast_threadpool_options options;
1799 if (pj_init() != PJ_SUCCESS) {
1800 return AST_MODULE_LOAD_DECLINE;
1803 if (pjlib_util_init() != PJ_SUCCESS) {
1805 return AST_MODULE_LOAD_DECLINE;
1808 pj_caching_pool_init(&caching_pool, NULL, 1024 * 1024);
1809 if (pjsip_endpt_create(&caching_pool.factory, "SIP", &ast_pjsip_endpoint) != PJ_SUCCESS) {
1810 ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n");
1814 /* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that,
1815 * we need to stop PJSIP from doing it automatically
1817 remove_request_headers(ast_pjsip_endpoint);
1819 memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL);
1821 ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n");
1825 if (ast_sip_initialize_system()) {
1826 ast_log(LOG_ERROR, "Failed to initialize SIP system configuration. Aborting load\n");
1830 sip_get_threadpool_options(&options);
1831 options.thread_start = sip_thread_start;
1832 sip_threadpool = ast_threadpool_create("SIP", NULL, &options);
1833 if (!sip_threadpool) {
1834 ast_log(LOG_ERROR, "Failed to create SIP threadpool. Aborting load\n");
1838 pjsip_tsx_layer_init_module(ast_pjsip_endpoint);
1839 pjsip_ua_init_module(ast_pjsip_endpoint, NULL);
1841 monitor_continue = 1;
1842 status = pj_thread_create(memory_pool, "SIP", (pj_thread_proc *) &monitor_thread_exec,
1843 NULL, PJ_THREAD_DEFAULT_STACK_SIZE * 2, 0, &monitor_thread);
1844 if (status != PJ_SUCCESS) {
1845 ast_log(LOG_ERROR, "Failed to start SIP monitor thread. Aborting load\n");
1849 ast_sip_initialize_global_headers();
1851 if (ast_res_pjsip_initialize_configuration()) {
1852 ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
1856 if (ast_sip_initialize_distributor()) {
1857 ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n");
1861 if (ast_sip_initialize_outbound_authentication()) {
1862 ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n");
1866 ast_res_pjsip_init_options_handling(0);
1868 ast_res_pjsip_init_contact_transports();
1870 return AST_MODULE_LOAD_SUCCESS;
1873 ast_sip_destroy_distributor();
1874 ast_res_pjsip_destroy_configuration();
1875 ast_sip_destroy_global_headers();
1876 if (monitor_thread) {
1877 stop_monitor_thread();
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);
1888 return AST_MODULE_LOAD_DECLINE;
1891 static int reload_module(void)
1893 if (ast_res_pjsip_reload_configuration()) {
1894 return AST_MODULE_LOAD_DECLINE;
1896 ast_res_pjsip_init_options_handling(1);
1900 static int unload_pjsip(void *data)
1903 pj_pool_release(memory_pool);
1906 if (ast_pjsip_endpoint) {
1907 pjsip_endpt_destroy(ast_pjsip_endpoint);
1908 ast_pjsip_endpoint = NULL;
1910 pj_caching_pool_destroy(&caching_pool);
1914 static int unload_module(void)
1916 ast_res_pjsip_cleanup_options_handling();
1917 ast_sip_destroy_distributor();
1918 ast_res_pjsip_destroy_configuration();
1919 ast_sip_destroy_global_headers();
1920 if (monitor_thread) {
1921 stop_monitor_thread();
1923 /* The thread this is called from cannot call PJSIP/PJLIB functions,
1924 * so we have to push the work to the threadpool to handle
1926 ast_sip_push_task_synchronous(NULL, unload_pjsip, NULL);
1928 ast_threadpool_shutdown(sip_threadpool);
1933 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Basic SIP resource",
1934 .load = load_module,
1935 .unload = unload_module,
1936 .reload = reload_module,
1937 .load_pri = AST_MODPRI_CHANNEL_DEPEND - 5,
projects / asterisk/asterisk.git / blob