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Diffstat (limited to 'res/res_pjsip.c')
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diff --git a/res/res_pjsip.c b/res/res_pjsip.c new file mode 100644 index 000000000..2326c9add --- /dev/null +++ b/res/res_pjsip.c @@ -0,0 +1,1900 @@ +/* + * Asterisk -- An open source telephony toolkit. + * + * Copyright (C) 2013, Digium, Inc. + * + * Mark Michelson <mmichelson@digium.com> + * + * See http://www.asterisk.org for more information about + * the Asterisk project. Please do not directly contact + * any of the maintainers of this project for assistance; + * the project provides a web site, mailing lists and IRC + * channels for your use. + * + * This program is free software, distributed under the terms of + * the GNU General Public License Version 2. See the LICENSE file + * at the top of the source tree. + */ + +#include "asterisk.h" + +#include <pjsip.h> +/* Needed for SUBSCRIBE, NOTIFY, and PUBLISH method definitions */ +#include <pjsip_simple.h> +#include <pjlib.h> + +#include "asterisk/res_pjsip.h" +#include "res_pjsip/include/res_pjsip_private.h" +#include "asterisk/linkedlists.h" +#include "asterisk/logger.h" +#include "asterisk/lock.h" +#include "asterisk/utils.h" +#include "asterisk/astobj2.h" +#include "asterisk/module.h" +#include "asterisk/threadpool.h" +#include "asterisk/taskprocessor.h" +#include "asterisk/uuid.h" +#include "asterisk/sorcery.h" + +/*** MODULEINFO + <depend>pjproject</depend> + <depend>res_sorcery_config</depend> + <support_level>core</support_level> + ***/ + +/*** DOCUMENTATION + <configInfo name="res_pjsip" language="en_US"> + <synopsis>SIP Resource using PJProject</synopsis> + <configFile name="pjsip.conf"> + <configObject name="endpoint"> + <synopsis>Endpoint</synopsis> + <description><para> + The <emphasis>Endpoint</emphasis> is the primary configuration object. + It contains the core SIP related options only, endpoints are <emphasis>NOT</emphasis> + dialable entries of their own. Communication with another SIP device is + accomplished via Addresses of Record (AoRs) which have one or more + contacts assicated with them. Endpoints <emphasis>NOT</emphasis> configured to + use a <literal>transport</literal> will default to first transport found + in <filename>pjsip.conf</filename> that matches its type. + </para> + <para>Example: An Endpoint has been configured with no transport. + When it comes time to call an AoR, PJSIP will find the + first transport that matches the type. A SIP URI of <literal>sip:5000@[11::33]</literal> + will use the first IPv6 transport and try to send the request. + </para> + <para>If the anonymous endpoint identifier is in use an endpoint with the name + "anonymous@domain" will be searched for as a last resort. If this is not found + it will fall back to searching for "anonymous". If neither endpoints are found + the anonymous endpoint identifier will not return an endpoint and anonymous + calling will not be possible. + </para> + </description> + <configOption name="100rel" default="yes"> + <synopsis>Allow support for RFC3262 provisional ACK tags</synopsis> + <description> + <enumlist> + <enum name="no" /> + <enum name="required" /> + <enum name="yes" /> + </enumlist> + </description> + </configOption> + <configOption name="aggregate_mwi" default="yes"> + <synopsis></synopsis> + <description><para>When enabled, <replaceable>aggregate_mwi</replaceable> condenses message + waiting notifications from multiple mailboxes into a single NOTIFY. If it is disabled, + individual NOTIFYs are sent for each mailbox.</para></description> + </configOption> + <configOption name="allow"> + <synopsis>Media Codec(s) to allow</synopsis> + </configOption> + <configOption name="aors"> + <synopsis>AoR(s) to be used with the endpoint</synopsis> + <description><para> + List of comma separated AoRs that the endpoint should be associated with. + </para></description> + </configOption> + <configOption name="auth"> + <synopsis>Authentication Object(s) associated with the endpoint</synopsis> + <description><para> + This is a comma-delimited list of <replaceable>auth</replaceable> sections defined + in <filename>pjsip.conf</filename> to be used to verify inbound connection attempts. + </para><para> + Endpoints without an <literal>authentication</literal> object + configured will allow connections without vertification. + </para></description> + </configOption> + <configOption name="callerid"> + <synopsis>CallerID information for the endpoint</synopsis> + <description><para> + Must be in the format <literal>Name <Number></literal>, + or only <literal><Number></literal>. + </para></description> + </configOption> + <configOption name="callerid_privacy"> + <synopsis>Default privacy level</synopsis> + <description> + <enumlist> + <enum name="allowed_not_screened" /> + <enum name="allowed_passed_screened" /> + <enum name="allowed_failed_screened" /> + <enum name="allowed" /> + <enum name="prohib_not_screened" /> + <enum name="prohib_passed_screened" /> + <enum name="prohib_failed_screened" /> + <enum name="prohib" /> + <enum name="unavailable" /> + </enumlist> + </description> + </configOption> + <configOption name="callerid_tag"> + <synopsis>Internal id_tag for the endpoint</synopsis> + </configOption> + <configOption name="context"> + <synopsis>Dialplan context for inbound sessions</synopsis> + </configOption> + <configOption name="direct_media_glare_mitigation" default="none"> + <synopsis>Mitigation of direct media (re)INVITE glare</synopsis> + <description> + <para> + This setting attempts to avoid creating INVITE glare scenarios + by disabling direct media reINVITEs in one direction thereby allowing + designated servers (according to this option) to initiate direct + media reINVITEs without contention and significantly reducing call + setup time. + </para> + <para> + A more detailed description of how this option functions can be found on + the Asterisk wiki https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance + </para> + <enumlist> + <enum name="none" /> + <enum name="outgoing" /> + <enum name="incoming" /> + </enumlist> + </description> + </configOption> + <configOption name="direct_media_method" default="invite"> + <synopsis>Direct Media method type</synopsis> + <description> + <para>Method for setting up Direct Media between endpoints.</para> + <enumlist> + <enum name="invite" /> + <enum name="reinvite"> + <para>Alias for the <literal>invite</literal> value.</para> + </enum> + <enum name="update" /> + </enumlist> + </description> + </configOption> + <configOption name="connected_line_method" default="invite"> + <synopsis>Connected line method type</synopsis> + <description> + <para>Method used when updating connected line information.</para> + <enumlist> + <enum name="invite" /> + <enum name="reinvite"> + <para>Alias for the <literal>invite</literal> value.</para> + </enum> + <enum name="update" /> + </enumlist> + </description> + </configOption> + <configOption name="direct_media" default="yes"> + <synopsis>Determines whether media may flow directly between endpoints.</synopsis> + </configOption> + <configOption name="disable_direct_media_on_nat" default="no"> + <synopsis>Disable direct media session refreshes when NAT obstructs the media session</synopsis> + </configOption> + <configOption name="disallow"> + <synopsis>Media Codec(s) to disallow</synopsis> + </configOption> + <configOption name="dtmfmode" default="rfc4733"> + <synopsis>DTMF mode</synopsis> + <description> + <para>This setting allows to choose the DTMF mode for endpoint communication.</para> + <enumlist> + <enum name="rfc4733"> + <para>DTMF is sent out of band of the main audio stream.This + supercedes the older <emphasis>RFC-2833</emphasis> used within + the older <literal>chan_sip</literal>.</para> + </enum> + <enum name="inband"> + <para>DTMF is sent as part of audio stream.</para> + </enum> + <enum name="info"> + <para>DTMF is sent as SIP INFO packets.</para> + </enum> + </enumlist> + </description> + </configOption> + <configOption name="external_media_address"> + <synopsis>IP used for External Media handling</synopsis> + </configOption> + <configOption name="force_rport" default="yes"> + <synopsis>Force use of return port</synopsis> + </configOption> + <configOption name="ice_support" default="no"> + <synopsis>Enable the ICE mechanism to help traverse NAT</synopsis> + </configOption> + <configOption name="identify_by" default="username,location"> + <synopsis>Way(s) for Endpoint to be identified</synopsis> + <description><para> + There are currently two methods to identify an endpoint. By default + both are used to identify an endpoint. + </para> + <enumlist> + <enum name="username" /> + <enum name="location" /> + <enum name="username,location" /> + </enumlist> + </description> + </configOption> + <configOption name="mailboxes"> + <synopsis>Mailbox(es) to be associated with</synopsis> + </configOption> + <configOption name="mohsuggest" default="default"> + <synopsis>Default Music On Hold class</synopsis> + </configOption> + <configOption name="outbound_auth"> + <synopsis>Authentication object used for outbound requests</synopsis> + </configOption> + <configOption name="outbound_proxy"> + <synopsis>Proxy through which to send requests</synopsis> + </configOption> + <configOption name="rewrite_contact"> + <synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis> + </configOption> + <configOption name="rtp_ipv6" default="no"> + <synopsis>Allow use of IPv6 for RTP traffic</synopsis> + </configOption> + <configOption name="rtp_symmetric" default="no"> + <synopsis>Enforce that RTP must be symmetric</synopsis> + </configOption> + <configOption name="send_pai" default="no"> + <synopsis>Send the P-Asserted-Identity header</synopsis> + </configOption> + <configOption name="send_rpid" default="no"> + <synopsis>Send the Remote-Party-ID header</synopsis> + </configOption> + <configOption name="timers_min_se" default="90"> + <synopsis>Minimum session timers expiration period</synopsis> + <description><para> + Minimium session timer expiration period. Time in seconds. + </para></description> + </configOption> + <configOption name="timers" default="yes"> + <synopsis>Session timers for SIP packets</synopsis> + <description> + <enumlist> + <enum name="forced" /> + <enum name="no" /> + <enum name="required" /> + <enum name="yes" /> + </enumlist> + </description> + </configOption> + <configOption name="timers_sess_expires" default="1800"> + <synopsis>Maximum session timer expiration period</synopsis> + <description><para> + Maximium session timer expiration period. Time in seconds. + </para></description> + </configOption> + <configOption name="transport"> + <synopsis>Desired transport configuration</synopsis> + <description><para> + This will set the desired transport configuration to send SIP data through. + </para> + <warning><para>Not specifying a transport will <emphasis>DEFAULT</emphasis> + to the first configured transport in <filename>pjsip.conf</filename> which is + valid for the URI we are trying to contact. + </para></warning> + </description> + </configOption> + <configOption name="trust_id_inbound" default="no"> + <synopsis>Accept identification information received from this endpoint</synopsis> + <description><para>This option determines whether Asterisk will accept + identification from the endpoint from headers such as P-Asserted-Identity + or Remote-Party-ID header. This option applies both to calls originating from the + endpoint and calls originating from Asterisk. If <literal>no</literal>, the + configured Caller-ID from pjsip.conf will always be used as the identity for + the endpoint.</para></description> + </configOption> + <configOption name="trust_id_outbound" default="no"> + <synopsis>Send private identification details to the endpoint.</synopsis> + <description><para>This option determines whether res_pjsip will send private + identification information to the endpoint. If <literal>no</literal>, + private Caller-ID information will not be forwarded to the endpoint. + "Private" in this case refers to any method of restricting identification. + Example: setting <replaceable>callerid_privacy</replaceable> to any + <literal>prohib</literal> variation. + Example: If <replaceable>trust_id_inbound</replaceable> is set to + <literal>yes</literal>, the presence of a <literal>Privacy: id</literal> + header in a SIP request or response would indicate the identification + provided in the request is private.</para></description> + </configOption> + <configOption name="type"> + <synopsis>Must be of type 'endpoint'.</synopsis> + </configOption> + <configOption name="use_ptime" default="no"> + <synopsis>Use Endpoint's requested packetisation interval</synopsis> + </configOption> + <configOption name="use_avpf" default="no"> + <synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this + endpoint.</synopsis> + <description><para> + If set to <literal>yes</literal>, res_pjsip will use use the AVPF or SAVPF RTP + profile for all media offers on outbound calls and media updates and will + decline media offers not using the AVPF or SAVPF profile. + </para><para> + If set to <literal>no</literal>, res_pjsip will use use the AVP or SAVP RTP + profile for all media offers on outbound calls and media updates and will + decline media offers not using the AVP or SAVP profile. + </para></description> + </configOption> + <configOption name="media_encryption" default="no"> + <synopsis>Determines whether res_pjsip will use and enforce usage of media encryption + for this endpoint.</synopsis> + <description> + <enumlist> + <enum name="no"><para> + res_pjsip will offer no encryption and allow no encryption to be setup. + </para></enum> + <enum name="sdes"><para> + res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP + transport should be used in conjunction with this option to prevent + exposure of media encryption keys. + </para></enum> + <enum name="dtls"><para> + res_pjsip will offer DTLS-SRTP setup. + </para></enum> + </enumlist> + </description> + </configOption> + <configOption name="inband_progress" default="no"> + <synopsis>Determines whether chan_pjsip will indicate ringing using inband + progress.</synopsis> + <description><para> + If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress + when told to indicate ringing and will immediately start sending ringing + as audio. + </para><para> + If set to <literal>no</literal>, chan_pjsip will send a 180 Ringing when told + to indicate ringing and will NOT send it as audio. + </para></description> + </configOption> + <configOption name="callgroup"> + <synopsis>The numeric pickup groups for a channel.</synopsis> + <description><para> + Can be set to a comma separated list of numbers or ranges between the values + of 0-63 (maximum of 64 groups). + </para></description> + </configOption> + <configOption name="pickupgroup"> + <synopsis>The numeric pickup groups that a channel can pickup.</synopsis> + <description><para> + Can be set to a comma separated list of numbers or ranges between the values + of 0-63 (maximum of 64 groups). + </para></description> + </configOption> + <configOption name="namedcallgroup"> + <synopsis>The named pickup groups for a channel.</synopsis> + <description><para> + Can be set to a comma separated list of case sensitive strings limited by + supported line length. + </para></description> + </configOption> + <configOption name="namedpickupgroup"> + <synopsis>The named pickup groups that a channel can pickup.</synopsis> + <description><para> + Can be set to a comma separated list of case sensitive strings limited by + supported line length. + </para></description> + </configOption> + <configOption name="devicestate_busy_at" default="0"> + <synopsis>The number of in-use channels which will cause busy to be returned as device state</synopsis> + <description><para> + When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the + PJSIP channel driver will return busy as the device state instead of in use. + </para></description> + </configOption> + <configOption name="t38udptl" default="no"> + <synopsis>Whether T.38 UDPTL support is enabled or not</synopsis> + <description><para> + If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted + and relayed. + </para></description> + </configOption> + <configOption name="t38udptl_ec" default="none"> + <synopsis>T.38 UDPTL error correction method</synopsis> + <description> + <enumlist> + <enum name="none"><para> + No error correction should be used. + </para></enum> + <enum name="fec"><para> + Forward error correction should be used. + </para></enum> + <enum name="redundancy"><para> + Redundacy error correction should be used. + </para></enum> + </enumlist> + </description> + </configOption> + <configOption name="t38udptl_maxdatagram" default="0"> + <synopsis>T.38 UDPTL maximum datagram size</synopsis> + <description><para> + This option can be set to override the maximum datagram of a remote endpoint for broken + endpoints. + </para></description> + </configOption> + <configOption name="faxdetect" default="no"> + <synopsis>Whether CNG tone detection is enabled</synopsis> + <description><para> + This option can be set to send the session to the fax extension when a CNG tone is + detected. + </para></description> + </configOption> + <configOption name="t38udptl_nat" default="no"> + <synopsis>Whether NAT support is enabled on UDPTL sessions</synopsis> + <description><para> + When enabled the UDPTL stack will send UDPTL packets to the source address of + received packets. + </para></description> + </configOption> + <configOption name="t38udptl_ipv6" default="no"> + <synopsis>Whether IPv6 is used for UDPTL Sessions</synopsis> + <description><para> + When enabled the UDPTL stack will use IPv6. + </para></description> + </configOption> + <configOption name="tonezone"> + <synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis> + </configOption> + <configOption name="language"> + <synopsis>Set the default language to use for channels created for this endpoint.</synopsis> + </configOption> + <configOption name="one_touch_recording" default="no"> + <synopsis>Determines whether one-touch recording is allowed for this endpoint.</synopsis> + <see-also> + <ref type="configOption">recordonfeature</ref> + <ref type="configOption">recordofffeature</ref> + </see-also> + </configOption> + <configOption name="recordonfeature" default="automixmon"> + <synopsis>The feature to enact when one-touch recording is turned on.</synopsis> + <description> + <para>When an INFO request for one-touch recording arrives with a Record header set to "on", this + feature will be enabled for the channel. The feature designated here can be any built-in + or dynamic feature defined in features.conf.</para> + <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note> + </description> + <see-also> + <ref type="configOption">one_touch_recording</ref> + <ref type="configOption">recordofffeature</ref> + </see-also> + </configOption> + <configOption name="recordofffeature" default="automixmon"> + <synopsis>The feature to enact when one-touch recording is turned off.</synopsis> + <description> + <para>When an INFO request for one-touch recording arrives with a Record header set to "off", this + feature will be enabled for the channel. The feature designated here can be any built-in + or dynamic feature defined in features.conf.</para> + <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note> + </description> + <see-also> + <ref type="configOption">one_touch_recording</ref> + <ref type="configOption">recordonfeature</ref> + </see-also> + </configOption> + <configOption name="rtpengine" default="asterisk"> + <synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis> + </configOption> + <configOption name="allowtransfer" default="yes"> + <synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis> + </configOption> + <configOption name="sdpowner" default="-"> + <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis> + </configOption> + <configOption name="sdpsession" default="Asterisk"> + <synopsis>String used for the SDP session (s=) line.</synopsis> + </configOption> + <configOption name="tos_audio"> + <synopsis>DSCP TOS bits for audio streams</synopsis> + <description><para> + See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings + </para></description> + </configOption> + <configOption name="tos_video"> + <synopsis>DSCP TOS bits for video streams</synopsis> + <description><para> + See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings + </para></description> + </configOption> + <configOption name="cos_audio"> + <synopsis>Priority for audio streams</synopsis> + <description><para> + See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings + </para></description> + </configOption> + <configOption name="cos_video"> + <synopsis>Priority for video streams</synopsis> + <description><para> + See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings + </para></description> + </configOption> + <configOption name="allowsubscribe" default="yes"> + <synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis> + </configOption> + <configOption name="subminexpiry" default="60"> + <synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis> + </configOption> + <configOption name="fromuser"> + <synopsis>Username to use in From header for requests to this endpoint.</synopsis> + </configOption> + <configOption name="mwifromuser"> + <synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis> + </configOption> + <configOption name="fromdomain"> + <synopsis>Domain to user in From header for requests to this endpoint.</synopsis> + </configOption> + <configOption name="dtlsverify"> + <synopsis>Verify that the provided peer certificate is valid</synopsis> + <description><para> + This option only applies if <replaceable>media_encryption</replaceable> is + set to <literal>dtls</literal>. + </para></description> + </configOption> + <configOption name="dtlsrekey"> + <synopsis>Interval at which to renegotiate the TLS session and rekey the SRTP session</synopsis> + <description><para> + This option only applies if <replaceable>media_encryption</replaceable> is + set to <literal>dtls</literal>. + </para><para> + If this is not set or the value provided is 0 rekeying will be disabled. + </para></description> + </configOption> + <configOption name="dtlscertfile"> + <synopsis>Path to certificate file to present to peer</synopsis> + <description><para> + This option only applies if <replaceable>media_encryption</replaceable> is + set to <literal>dtls</literal>. + </para></description> + </configOption> + <configOption name="dtlsprivatekey"> + <synopsis>Path to private key for certificate file</synopsis> + <description><para> + This option only applies if <replaceable>media_encryption</replaceable> is + set to <literal>dtls</literal>. + </para></description> + </configOption> + <configOption name="dtlscipher"> + <synopsis>Cipher to use for DTLS negotiation</synopsis> + <description><para> + This option only applies if <replaceable>media_encryption</replaceable> is + set to <literal>dtls</literal>. + </para><para> + Many options for acceptable ciphers. See link for more: + http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS + </para></description> + </configOption> + <configOption name="dtlscafile"> + <synopsis>Path to certificate authority certificate</synopsis> + <description><para> + This option only applies if <replaceable>media_encryption</replaceable> is + set to <literal>dtls</literal>. + </para></description> + </configOption> + <configOption name="dtlscapath"> + <synopsis>Path to a directory containing certificate authority certificates</synopsis> + <description><para> + This option only applies if <replaceable>media_encryption</replaceable> is + set to <literal>dtls</literal>. + </para></description> + </configOption> + <configOption name="dtlssetup"> + <synopsis>Whether we are willing to accept connections, connect to the other party, or both.</synopsis> + <description> + <para> + This option only applies if <replaceable>media_encryption</replaceable> is + set to <literal>dtls</literal>. + </para> + <enumlist> + <enum name="active"><para> + res_pjsip will make a connection to the peer. + </para></enum> + <enum name="passive"><para> + res_pjsip will accept connections from the peer. + </para></enum> + <enum name="actpass"><para> + res_pjsip will offer and accept connections from the peer. + </para></enum> + </enumlist> + </description> + </configOption> + <configOption name="srtp_tag_32"> + <synopsis>Determines whether 32 byte tags should be used instead of 80 byte tags.</synopsis> + <description><para> + This option only applies if <replaceable>media_encryption</replaceable> is + set to <literal>sdes</literal> or <literal>dtls</literal>. + </para></description> + </configOption> + </configObject> + <configObject name="auth"> + <synopsis>Authentication type</synopsis> + <description><para> + Authentication objects hold the authenitcation information for use + by <literal>endpoints</literal>. This also allows for multiple <literal> + endpoints</literal> to use the same information. Choice of MD5/plaintext + and setting of username. + </para></description> + <configOption name="auth_type" default="userpass"> + <synopsis>Authentication type</synopsis> + <description><para> + This option specifies which of the password style config options should be read, + either 'password' or 'md5_cred' when trying to authenticate an endpoint inbound request. + </para> + <enumlist> + <enum name="md5"/> + <enum name="userpass"/> + </enumlist> + </description> + </configOption> + <configOption name="nonce_lifetime" default="32"> + <synopsis>Lifetime of a nonce associated with this authentication config.</synopsis> + </configOption> + <configOption name="md5_cred"> + <synopsis>MD5 Hash used for authentication.</synopsis> + <description><para>Only used when auth_type is <literal>md5</literal>.</para></description> + </configOption> + <configOption name="password"> + <synopsis>PlainText password used for authentication.</synopsis> + <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description> + </configOption> + <configOption name="realm" default="asterisk"> + <synopsis>SIP realm for endpoint</synopsis> + </configOption> + <configOption name="type"> + <synopsis>Must be 'auth'</synopsis> + </configOption> + <configOption name="username"> + <synopsis>Username to use for account</synopsis> + </configOption> + </configObject> + <configObject name="nat_hook"> + <synopsis>XXX This exists only to prevent XML documentation errors.</synopsis> + <configOption name="external_media_address"> + <synopsis>I should be undocumented or hidden</synopsis> + </configOption> + <configOption name="method"> + <synopsis>I should be undocumented or hidden</synopsis> + </configOption> + </configObject> + <configObject name="domain_alias"> + <synopsis>Domain Alias</synopsis> + <description><para> + Signifies that a domain is an alias. Used for checking the domain of + the AoR to which the endpoint is binding. + </para></description> + <configOption name="type"> + <synopsis>Must be of type 'domain_alias'.</synopsis> + </configOption> + <configOption name="domain"> + <synopsis>Domain to be aliased</synopsis> + </configOption> + </configObject> + <configObject name="transport"> + <synopsis>SIP Transport</synopsis> + <description><para> + <emphasis>Transports</emphasis> + </para> + <para>There are different transports and protocol derivatives + supported by <literal>res_pjsip</literal>. They are in order of + preference: UDP, TCP, and WebSocket (WS).</para> + <warning><para> + Multiple endpoints using the same connection is <emphasis>NOT</emphasis> + supported. Doing so may result in broken calls. + </para></warning> + </description> + <configOption name="async_operations" default="1"> + <synopsis>Number of simultaneous Asynchronous Operations</synopsis> + </configOption> + <configOption name="bind"> + <synopsis>IP Address and optional port to bind to for this transport</synopsis> + </configOption> + <configOption name="ca_list_file"> + <synopsis>File containing a list of certificates to read (TLS ONLY)</synopsis> + </configOption> + <configOption name="cert_file"> + <synopsis>Certificate file for endpoint (TLS ONLY)</synopsis> + </configOption> + <configOption name="cipher"> + <synopsis>Preferred Cryptography Cipher (TLS ONLY)</synopsis> + <description><para> + Many options for acceptable ciphers see link for more: + http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS + </para></description> + </configOption> + <configOption name="domain"> + <synopsis>Domain the transport comes from</synopsis> + </configOption> + <configOption name="external_media_address"> + <synopsis>External Address to use in RTP handling</synopsis> + </configOption> + <configOption name="external_signaling_address"> + <synopsis>External address for SIP signalling</synopsis> + </configOption> + <configOption name="external_signaling_port" default="0"> + <synopsis>External port for SIP signalling</synopsis> + </configOption> + <configOption name="method"> + <synopsis>Method of SSL transport (TLS ONLY)</synopsis> + <description> + <enumlist> + <enum name="default" /> + <enum name="unspecified" /> + <enum name="tlsv1" /> + <enum name="sslv2" /> + <enum name="sslv3" /> + <enum name="sslv23" /> + </enumlist> + </description> + </configOption> + <configOption name="localnet"> + <synopsis>Network to consider local (used for NAT purposes).</synopsis> + <description><para>This must be in CIDR or dotted decimal format with the IP + and mask separated with a slash ('/').</para></description> + </configOption> + <configOption name="password"> + <synopsis>Password required for transport</synopsis> + </configOption> + <configOption name="privkey_file"> + <synopsis>Private key file (TLS ONLY)</synopsis> + </configOption> + <configOption name="protocol" default="udp"> + <synopsis>Protocol to use for SIP traffic</synopsis> + <description> + <enumlist> + <enum name="udp" /> + <enum name="tcp" /> + <enum name="tls" /> + </enumlist> + </description> + </configOption> + <configOption name="require_client_cert" default="false"> + <synopsis>Require client certificate (TLS ONLY)</synopsis> + </configOption> + <configOption name="type"> + <synopsis>Must be of type 'transport'.</synopsis> + </configOption> + <configOption name="verify_client" default="false"> + <synopsis>Require verification of client certificate (TLS ONLY)</synopsis> + </configOption> + <configOption name="verify_server" default="false"> + <synopsis>Require verification of server certificate (TLS ONLY)</synopsis> + </configOption> + </configObject> + <configObject name="contact"> + <synopsis>A way of creating an aliased name to a SIP URI</synopsis> + <description><para> + Contacts are a way to hide SIP URIs from the dialplan directly. + They are also used to make a group of contactable parties when + in use with <literal>AoR</literal> lists. + </para></description> + <configOption name="type"> + <synopsis>Must be of type 'contact'.</synopsis> + </configOption> + <configOption name="uri"> + <synopsis>SIP URI to contact peer</synopsis> + </configOption> + <configOption name="expiration_time"> + <synopsis>Time to keep alive a contact</synopsis> + <description><para> + Time to keep alive a contact. String style specification. + </para></description> + </configOption> + <configOption name="qualify_frequency" default="0"> + <synopsis>Interval at which to qualify a contact</synopsis> + <description><para> + Interval between attempts to qualify the contact for reachability. + If <literal>0</literal> never qualify. Time in seconds. + </para></description> + </configOption> + </configObject> + <configObject name="contact_status"> + <synopsis>Status for a contact</synopsis> + <description><para> + The contact status keeps track of whether or not a contact is reachable + and how long it took to qualify the contact (round trip time). + </para></description> + <configOption name="status"> + <synopsis>A contact's status</synopsis> + <description> + <enumlist> + <enum name="AVAILABLE" /> + <enum name="UNAVAILABLE" /> + </enumlist> + </description> + </configOption> + <configOption name="rtt"> + <synopsis>Round trip time</synopsis> + <description><para> + The time, in microseconds, it took to qualify the contact. + </para></description> + </configOption> + </configObject> + <configObject name="aor"> + <synopsis>The configuration for a location of an endpoint</synopsis> + <description><para> + An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no + AoRs are specified, an endpoint will not be reachable by Asterisk. + Beyond that, an AoR has other uses within Asterisk. + </para><para> + An <literal>AoR</literal> is a way to allow dialing a group + of <literal>Contacts</literal> that all use the same + <literal>endpoint</literal> for calls. + </para><para> + This can be used as another way of grouping a list of contacts to dial + rather than specifing them each directly when dialing via the dialplan. + This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>. + </para></description> + <configOption name="contact"> + <synopsis>Permanent contacts assigned to AoR</synopsis> + <description><para> + Contacts included in this list will be called whenever referenced + by <literal>chan_pjsip</literal>. + </para></description> + </configOption> + <configOption name="default_expiration" default="3600"> + <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis> + </configOption> + <configOption name="mailboxes"> + <synopsis>Mailbox(es) to be associated with</synopsis> + <description><para>This option applies when an external entity subscribes to an AoR + for message waiting indications. The mailboxes specified here will be + subscribed to.</para></description> + </configOption> + <configOption name="maximum_expiration" default="7200"> + <synopsis>Maximum time to keep an AoR</synopsis> + <description><para> + Maximium time to keep a peer with explicit expiration. Time in seconds. + </para></description> + </configOption> + <configOption name="max_contacts" default="0"> + <synopsis>Maximum number of contacts that can bind to an AoR</synopsis> + <description><para> + Maximum number of contacts that can associate with this AoR. + </para> + <note><para>This should be set to <literal>1</literal> and + <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you + wish to stick with the older <literal>chan_sip</literal> behaviour. + </para></note> + </description> + </configOption> + <configOption name="minimum_expiration" default="60"> + <synopsis>Minimum keep alive time for an AoR</synopsis> + <description><para> + Minimum time to keep a peer with an explict expiration. Time in seconds. + </para></description> + </configOption> + <configOption name="remove_existing" default="no"> + <synopsis>Determines whether new contacts replace existing ones.</synopsis> + <description><para> + On receiving a new registration to the AoR should it remove + the existing contact that was registered against it? + </para> + <note><para>This should be set to <literal>yes</literal> and + <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you + wish to stick with the older <literal>chan_sip</literal> behaviour. + </para></note> + </description> + </configOption> + <configOption name="type"> + <synopsis>Must be of type 'aor'.</synopsis> + </configOption> + <configOption name="qualify_frequency" default="0"> + <synopsis>Interval at which to qualify an AoR</synopsis> + <description><para> + Interval between attempts to qualify the AoR for reachability. + If <literal>0</literal> never qualify. Time in seconds. + </para></description> + </configOption> + <configOption name="authenticate_qualify" default="no"> + <synopsis>Authenticates a qualify request if needed</synopsis> + <description><para> + If true and a qualify request receives a challenge or authenticate response + authentication is attempted before declaring the contact available. + </para></description> + </configOption> + </configObject> + <configObject name="system"> + <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis> + <description><para> + The settings in this section are global. In addition to being global, the values will + not be re-evaluated when a reload is performed. This is because the values must be set + before the SIP stack is initialized. The only way to reset these values is to either + restart Asterisk, or unload res_pjsip.so and then load it again. + </para></description> + <configOption name="timert1" default="500"> + <synopsis>Set transaction timer T1 value (milliseconds).</synopsis> + <description><para> + Timer T1 is the base for determining how long to wait before retransmitting + requests that receive no response when using an unreliable transport (e.g. UDP). + For more information on this timer, see RFC 3261, Section 17.1.1.1. + </para></description> + </configOption> + <configOption name="timerb" default="32000"> + <synopsis>Set transaction timer B value (milliseconds).</synopsis> + <description><para> + Timer B determines the maximum amount of time to wait after sending an INVITE + request before terminating the transaction. It is recommended that this be set + to 64 * Timer T1, but it may be set higher if desired. For more information on + this timer, see RFC 3261, Section 17.1.1.1. + </para></description> + </configOption> + <configOption name="compactheaders" default="no"> + <synopsis>Use the short forms of common SIP header names.</synopsis> + </configOption> + </configObject> + <configObject name="global"> + <synopsis>Options that apply globally to all SIP communications</synopsis> + <description><para> + The settings in this section are global. Unlike options in the <literal>system</literal> + section, these options can be refreshed by performing a reload. + </para></description> + <configOption name="maxforwards" default="70"> + <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis> + </configOption> + <configOption name="useragent" default="Asterisk <Asterisk Version>"> + <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis> + </configOption> + </configObject> + </configFile> + </configInfo> + ***/ + + +static pjsip_endpoint *ast_pjsip_endpoint; + +static struct ast_threadpool *sip_threadpool; + +static int register_service(void *data) +{ + pjsip_module **module = data; + if (!ast_pjsip_endpoint) { + ast_log(LOG_ERROR, "There is no PJSIP endpoint. Unable to register services\n"); + return -1; + } + if (pjsip_endpt_register_module(ast_pjsip_endpoint, *module) != PJ_SUCCESS) { + ast_log(LOG_ERROR, "Unable to register module %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name)); + return -1; + } + ast_debug(1, "Registered SIP service %.*s (%p)\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name), *module); + ast_module_ref(ast_module_info->self); + return 0; +} + +int ast_sip_register_service(pjsip_module *module) +{ + return ast_sip_push_task_synchronous(NULL, register_service, &module); +} + +static int unregister_service(void *data) +{ + pjsip_module **module = data; + ast_module_unref(ast_module_info->self); + if (!ast_pjsip_endpoint) { + return -1; + } + pjsip_endpt_unregister_module(ast_pjsip_endpoint, *module); + ast_debug(1, "Unregistered SIP service %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name)); + return 0; +} + +void ast_sip_unregister_service(pjsip_module *module) +{ + ast_sip_push_task_synchronous(NULL, unregister_service, &module); +} + +static struct ast_sip_authenticator *registered_authenticator; + +int ast_sip_register_authenticator(struct ast_sip_authenticator *auth) +{ + if (registered_authenticator) { + ast_log(LOG_WARNING, "Authenticator %p is already registered. Cannot register a new one\n", registered_authenticator); + return -1; + } + registered_authenticator = auth; + ast_debug(1, "Registered SIP authenticator module %p\n", auth); + ast_module_ref(ast_module_info->self); + return 0; +} + +void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth) +{ + if (registered_authenticator != auth) { + ast_log(LOG_WARNING, "Trying to unregister authenticator %p but authenticator %p registered\n", + auth, registered_authenticator); + return; + } + registered_authenticator = NULL; + ast_debug(1, "Unregistered SIP authenticator %p\n", auth); + ast_module_unref(ast_module_info->self); +} + +int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata) +{ + if (!registered_authenticator) { + ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is not required\n"); + return 0; + } + + return registered_authenticator->requires_authentication(endpoint, rdata); +} + +enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint, + pjsip_rx_data *rdata, pjsip_tx_data *tdata) +{ + if (!registered_authenticator) { + ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is successful\n"); + return 0; + } + return registered_authenticator->check_authentication(endpoint, rdata, tdata); +} + +static struct ast_sip_outbound_authenticator *registered_outbound_authenticator; + +int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *auth) +{ + if (registered_outbound_authenticator) { + ast_log(LOG_WARNING, "Outbound authenticator %p is already registered. Cannot register a new one\n", registered_outbound_authenticator); + return -1; + } + registered_outbound_authenticator = auth; + ast_debug(1, "Registered SIP outbound authenticator module %p\n", auth); + ast_module_ref(ast_module_info->self); + return 0; +} + +void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth) +{ + if (registered_outbound_authenticator != auth) { + ast_log(LOG_WARNING, "Trying to unregister outbound authenticator %p but outbound authenticator %p registered\n", + auth, registered_outbound_authenticator); + return; + } + registered_outbound_authenticator = NULL; + ast_debug(1, "Unregistered SIP outbound authenticator %p\n", auth); + ast_module_unref(ast_module_info->self); +} + +int ast_sip_create_request_with_auth(const struct ast_sip_auth_array *auths, pjsip_rx_data *challenge, + pjsip_transaction *tsx, pjsip_tx_data **new_request) +{ + if (!registered_outbound_authenticator) { + ast_log(LOG_WARNING, "No SIP outbound authenticator registered. Cannot respond to authentication challenge\n"); + return -1; + } + return registered_outbound_authenticator->create_request_with_auth(auths, challenge, tsx, new_request); +} + +struct endpoint_identifier_list { + struct ast_sip_endpoint_identifier *identifier; + AST_RWLIST_ENTRY(endpoint_identifier_list) list; +}; + +static AST_RWLIST_HEAD_STATIC(endpoint_identifiers, endpoint_identifier_list); + +int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier) +{ + struct endpoint_identifier_list *id_list_item; + SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK); + + id_list_item = ast_calloc(1, sizeof(*id_list_item)); + if (!id_list_item) { + ast_log(LOG_ERROR, "Unabled to add endpoint identifier. Out of memory.\n"); + return -1; + } + id_list_item->identifier = identifier; + + AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list); + ast_debug(1, "Registered endpoint identifier %p\n", identifier); + + ast_module_ref(ast_module_info->self); + return 0; +} + +void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier) +{ + struct endpoint_identifier_list *iter; + SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK); + AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) { + if (iter->identifier == identifier) { + AST_RWLIST_REMOVE_CURRENT(list); + ast_free(iter); + ast_debug(1, "Unregistered endpoint identifier %p\n", identifier); + ast_module_unref(ast_module_info->self); + break; + } + } + AST_RWLIST_TRAVERSE_SAFE_END; +} + +struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata) +{ + struct endpoint_identifier_list *iter; + struct ast_sip_endpoint *endpoint = NULL; + SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK); + AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) { + ast_assert(iter->identifier->identify_endpoint != NULL); + endpoint = iter->identifier->identify_endpoint(rdata); + if (endpoint) { + break; + } + } + return endpoint; +} + +pjsip_endpoint *ast_sip_get_pjsip_endpoint(void) +{ + return ast_pjsip_endpoint; +} + +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) +{ + pj_str_t tmp, local_addr; + pjsip_uri *uri; + pjsip_sip_uri *sip_uri; + pjsip_transport_type_e type = PJSIP_TRANSPORT_UNSPECIFIED; + int local_port; + char uuid_str[AST_UUID_STR_LEN]; + + if (ast_strlen_zero(user)) { + RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr); + if (!uuid) { + return -1; + } + user = ast_uuid_to_str(uuid, uuid_str, sizeof(uuid_str)); + } + + /* Parse the provided target URI so we can determine what transport it will end up using */ + pj_strdup_with_null(pool, &tmp, target); + + if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) || + (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) { + return -1; + } + + sip_uri = pjsip_uri_get_uri(uri); + + /* Determine the transport type to use */ + if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) { + type = PJSIP_TRANSPORT_TLS; + } else if (!sip_uri->transport_param.slen) { + type = PJSIP_TRANSPORT_UDP; + } else { + type = pjsip_transport_get_type_from_name(&sip_uri->transport_param); + } + + if (type == PJSIP_TRANSPORT_UNSPECIFIED) { + return -1; + } + + /* If the host is IPv6 turn the transport into an IPv6 version */ + if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) { + type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6); + } + + if (!ast_strlen_zero(domain)) { + from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE); + from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE, + "<%s:%s@%s%s%s>", + (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip", + user, + domain, + (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "", + (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : ""); + return 0; + } + + /* Get the local bound address for the transport that will be used when communicating with the provided URI */ + if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), pool, type, selector, + &local_addr, &local_port) != PJ_SUCCESS) { + return -1; + } + + /* If IPv6 was specified in the transport, set the proper type */ + if (pj_strchr(&local_addr, ':') && type < PJSIP_TRANSPORT_START_OTHER) { + type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6); + } + + from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE); + from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE, + "<%s:%s@%s%.*s%s:%d%s%s>", + (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip", + user, + (type & PJSIP_TRANSPORT_IPV6) ? "[" : "", + (int)local_addr.slen, + local_addr.ptr, + (type & PJSIP_TRANSPORT_IPV6) ? "]" : "", + local_port, + (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "", + (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : ""); + + return 0; +} + +static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector) +{ + RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup); + const char *transport_name = endpoint->transport; + + if (ast_strlen_zero(transport_name)) { + return 0; + } + + transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name); + + if (!transport || !transport->state) { + return -1; + } + + if (transport->state->transport) { + selector->type = PJSIP_TPSELECTOR_TRANSPORT; + selector->u.transport = transport->state->transport; + } else if (transport->state->factory) { + selector->type = PJSIP_TPSELECTOR_LISTENER; + selector->u.listener = transport->state->factory; + } else { + return -1; + } + + return 0; +} + +static int sip_get_tpselector_from_uri(const char *uri, pjsip_tpselector *selector) +{ + RAII_VAR(struct ast_sip_contact_transport *, contact_transport, NULL, ao2_cleanup); + + contact_transport = ast_sip_location_retrieve_contact_transport_by_uri(uri); + + if (!contact_transport) { + return -1; + } + + selector->type = PJSIP_TPSELECTOR_TRANSPORT; + selector->u.transport = contact_transport->transport; + + return 0; +} + +pjsip_dialog *ast_sip_create_dialog(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user) +{ + pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri; + pjsip_dialog *dlg = NULL; + const char *outbound_proxy = endpoint->outbound_proxy; + pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, }; + static const pj_str_t HCONTACT = { "Contact", 7 }; + + pj_cstr(&remote_uri, uri); + + if (pjsip_dlg_create_uac(pjsip_ua_instance(), &local_uri, NULL, &remote_uri, NULL, &dlg) != PJ_SUCCESS) { + return NULL; + } + + if (sip_get_tpselector_from_uri(uri, &selector) && sip_get_tpselector_from_endpoint(endpoint, &selector)) { + pjsip_dlg_terminate(dlg); + return NULL; + } + + if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) { + pjsip_dlg_terminate(dlg); + return NULL; + } + + /* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */ + pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri); + dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0); + dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL); + + /* If a request user has been specified and we are permitted to change it, do so */ + if (!ast_strlen_zero(request_user) && (PJSIP_URI_SCHEME_IS_SIP(dlg->target) || PJSIP_URI_SCHEME_IS_SIPS(dlg->target))) { + pjsip_sip_uri *target = pjsip_uri_get_uri(dlg->target); + pj_strdup2(dlg->pool, &target->user, request_user); + } + + /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */ + dlg->sess_count++; + + pjsip_dlg_set_transport(dlg, &selector); + + if (!ast_strlen_zero(outbound_proxy)) { + pjsip_route_hdr route_set, *route; + static const pj_str_t ROUTE_HNAME = { "Route", 5 }; + pj_str_t tmp; + + pj_list_init(&route_set); + + pj_strdup2_with_null(dlg->pool, &tmp, outbound_proxy); + if (!(route = pjsip_parse_hdr(dlg->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) { + pjsip_dlg_terminate(dlg); + return NULL; + } + pj_list_push_back(&route_set, route); + + pjsip_dlg_set_route_set(dlg, &route_set); + } + + dlg->sess_count--; + + return dlg; +} + +/* PJSIP doesn't know about the INFO method, so we have to define it ourselves */ +const pjsip_method pjsip_info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} }; +const pjsip_method pjsip_message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} }; + +static struct { + const char *method; + const pjsip_method *pmethod; +} methods [] = { + { "INVITE", &pjsip_invite_method }, + { "CANCEL", &pjsip_cancel_method }, + { "ACK", &pjsip_ack_method }, + { "BYE", &pjsip_bye_method }, + { "REGISTER", &pjsip_register_method }, + { "OPTIONS", &pjsip_options_method }, + { "SUBSCRIBE", &pjsip_subscribe_method }, + { "NOTIFY", &pjsip_notify_method }, + { "PUBLISH", &pjsip_publish_method }, + { "INFO", &pjsip_info_method }, + { "MESSAGE", &pjsip_message_method }, +}; + +static const pjsip_method *get_pjsip_method(const char *method) +{ + int i; + for (i = 0; i < ARRAY_LEN(methods); ++i) { + if (!strcmp(method, methods[i].method)) { + return methods[i].pmethod; + } + } + return NULL; +} + +static int create_in_dialog_request(const pjsip_method *method, struct pjsip_dialog *dlg, pjsip_tx_data **tdata) +{ + if (pjsip_dlg_create_request(dlg, method, -1, tdata) != PJ_SUCCESS) { + ast_log(LOG_WARNING, "Unable to create in-dialog request.\n"); + return -1; + } + + return 0; +} + +static int create_out_of_dialog_request(const pjsip_method *method, struct ast_sip_endpoint *endpoint, + const char *uri, pjsip_tx_data **tdata) +{ + RAII_VAR(struct ast_sip_contact *, contact, NULL, ao2_cleanup); + pj_str_t remote_uri; + pj_str_t from; + pj_pool_t *pool; + pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, }; + + if (ast_strlen_zero(uri)) { + if (!endpoint) { + ast_log(LOG_ERROR, "An endpoint and/or uri must be specified\n"); + return -1; + } + + contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors); + if (!contact || ast_strlen_zero(contact->uri)) { + ast_log(LOG_ERROR, "Unable to retrieve contact for endpoint %s\n", + ast_sorcery_object_get_id(endpoint)); + return -1; + } + + pj_cstr(&remote_uri, contact->uri); + } else { + pj_cstr(&remote_uri, uri); + } + + if (endpoint) { + if (sip_get_tpselector_from_endpoint(endpoint, &selector)) { + ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n", + ast_sorcery_object_get_id(endpoint)); + return -1; + } + } + + pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256); + + if (!pool) { + ast_log(LOG_ERROR, "Unable to create PJLIB memory pool\n"); + return -1; + } + + if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL, + endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) { + ast_log(LOG_ERROR, "Unable to create From header for %.*s request to endpoint %s\n", + (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint)); + pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool); + return -1; + } + + if (pjsip_endpt_create_request(ast_sip_get_pjsip_endpoint(), method, &remote_uri, + &from, &remote_uri, &from, NULL, -1, NULL, tdata) != PJ_SUCCESS) { + ast_log(LOG_ERROR, "Unable to create outbound %.*s request to endpoint %s\n", + (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint)); + pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool); + return -1; + } + + /* We can release this pool since request creation copied all the necessary + * data into the outbound request's pool + */ + pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool); + return 0; +} + +int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg, + struct ast_sip_endpoint *endpoint, const char *uri, + pjsip_tx_data **tdata) +{ + const pjsip_method *pmethod = get_pjsip_method(method); + + if (!pmethod) { + ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method); + return -1; + } + + if (dlg) { + return create_in_dialog_request(pmethod, dlg, tdata); + } else { + return create_out_of_dialog_request(pmethod, endpoint, uri, tdata); + } +} + +static int send_in_dialog_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg) +{ + if (pjsip_dlg_send_request(dlg, tdata, -1, NULL) != PJ_SUCCESS) { + ast_log(LOG_WARNING, "Unable to send in-dialog request.\n"); + return -1; + } + return 0; +} + +static void send_request_cb(void *token, pjsip_event *e) +{ + RAII_VAR(struct ast_sip_endpoint *, endpoint, token, ao2_cleanup); + pjsip_transaction *tsx = e->body.tsx_state.tsx; + pjsip_rx_data *challenge = e->body.tsx_state.src.rdata; + pjsip_tx_data *tdata; + + if (tsx->status_code != 401 && tsx->status_code != 407) { + return; + } + + if (!ast_sip_create_request_with_auth(&endpoint->outbound_auths, challenge, tsx, &tdata)) { + pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, NULL, NULL); + } +} + +static int send_out_of_dialog_request(pjsip_tx_data *tdata, struct ast_sip_endpoint *endpoint) +{ + ao2_ref(endpoint, +1); + if (pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, endpoint, send_request_cb) != PJ_SUCCESS) { + ast_log(LOG_ERROR, "Error attempting to send outbound %.*s request to endpoint %s\n", + (int) pj_strlen(&tdata->msg->line.req.method.name), + pj_strbuf(&tdata->msg->line.req.method.name), + ast_sorcery_object_get_id(endpoint)); + ao2_ref(endpoint, -1); + return -1; + } + + return 0; +} + +int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint) +{ + ast_assert(tdata->msg->type == PJSIP_REQUEST_MSG); + + if (dlg) { + return send_in_dialog_request(tdata, dlg); + } else { + return send_out_of_dialog_request(tdata, endpoint); + } +} + +int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value) +{ + pj_str_t hdr_name; + pj_str_t hdr_value; + pjsip_generic_string_hdr *hdr; + + pj_cstr(&hdr_name, name); + pj_cstr(&hdr_value, value); + + hdr = pjsip_generic_string_hdr_create(tdata->pool, &hdr_name, &hdr_value); + + pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) hdr); + return 0; +} + +static pjsip_msg_body *ast_body_to_pjsip_body(pj_pool_t *pool, const struct ast_sip_body *body) +{ + pj_str_t type; + pj_str_t subtype; + pj_str_t body_text; + + pj_cstr(&type, body->type); + pj_cstr(&subtype, body->subtype); + pj_cstr(&body_text, body->body_text); + + return pjsip_msg_body_create(pool, &type, &subtype, &body_text); +} + +int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body) +{ + pjsip_msg_body *pjsip_body = ast_body_to_pjsip_body(tdata->pool, body); + tdata->msg->body = pjsip_body; + return 0; +} + +int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies) +{ + int i; + /* NULL for type and subtype automatically creates "multipart/mixed" */ + pjsip_msg_body *body = pjsip_multipart_create(tdata->pool, NULL, NULL); + + for (i = 0; i < num_bodies; ++i) { + pjsip_multipart_part *part = pjsip_multipart_create_part(tdata->pool); + part->body = ast_body_to_pjsip_body(tdata->pool, bodies[i]); + pjsip_multipart_add_part(tdata->pool, body, part); + } + + tdata->msg->body = body; + return 0; +} + +int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text) +{ + size_t combined_size = strlen(body_text) + tdata->msg->body->len; + struct ast_str *body_buffer = ast_str_alloca(combined_size); + + ast_str_set(&body_buffer, 0, "%.*s%s", (int) tdata->msg->body->len, (char *) tdata->msg->body->data, body_text); + + tdata->msg->body->data = pj_pool_alloc(tdata->pool, combined_size); + pj_memcpy(tdata->msg->body->data, ast_str_buffer(body_buffer), combined_size); + tdata->msg->body->len = combined_size; + + return 0; +} + +struct ast_taskprocessor *ast_sip_create_serializer(void) +{ + struct ast_taskprocessor *serializer; + RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr); + char name[AST_UUID_STR_LEN]; + + if (!uuid) { + return NULL; + } + + ast_uuid_to_str(uuid, name, sizeof(name)); + + serializer = ast_threadpool_serializer(name, sip_threadpool); + if (!serializer) { + return NULL; + } + return serializer; +} + +int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data) +{ + if (serializer) { + return ast_taskprocessor_push(serializer, sip_task, task_data); + } else { + return ast_threadpool_push(sip_threadpool, sip_task, task_data); + } +} + +struct sync_task_data { + ast_mutex_t lock; + ast_cond_t cond; + int complete; + int fail; + int (*task)(void *); + void *task_data; +}; + +static int sync_task(void *data) +{ + struct sync_task_data *std = data; + std->fail = std->task(std->task_data); + + ast_mutex_lock(&std->lock); + std->complete = 1; + ast_cond_signal(&std->cond); + ast_mutex_unlock(&std->lock); + return std->fail; +} + +int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data) +{ + /* This method is an onion */ + struct sync_task_data std; + ast_mutex_init(&std.lock); + ast_cond_init(&std.cond, NULL); + std.fail = std.complete = 0; + std.task = sip_task; + std.task_data = task_data; + + if (serializer) { + if (ast_taskprocessor_push(serializer, sync_task, &std)) { + return -1; + } + } else { + if (ast_threadpool_push(sip_threadpool, sync_task, &std)) { + return -1; + } + } + + ast_mutex_lock(&std.lock); + while (!std.complete) { + ast_cond_wait(&std.cond, &std.lock); + } + ast_mutex_unlock(&std.lock); + + ast_mutex_destroy(&std.lock); + ast_cond_destroy(&std.cond); + return std.fail; +} + +void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size) +{ + size_t chars_to_copy = MIN(size - 1, pj_strlen(src)); + memcpy(dest, pj_strbuf(src), chars_to_copy); + dest[chars_to_copy] = '\0'; +} + +int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype) +{ + pjsip_media_type compare; + + if (!content_type) { + return 0; + } + + pjsip_media_type_init2(&compare, type, subtype); + + return pjsip_media_type_cmp(content_type, &compare, 0) ? -1 : 0; +} + +pj_caching_pool caching_pool; +pj_pool_t *memory_pool; +pj_thread_t *monitor_thread; +static int monitor_continue; + +static void *monitor_thread_exec(void *endpt) +{ + while (monitor_continue) { + const pj_time_val delay = {0, 10}; + pjsip_endpt_handle_events(ast_pjsip_endpoint, &delay); + } + return NULL; +} + +static void stop_monitor_thread(void) +{ + monitor_continue = 0; + pj_thread_join(monitor_thread); +} + +AST_THREADSTORAGE(pj_thread_storage); +AST_THREADSTORAGE(servant_id_storage); +#define SIP_SERVANT_ID 0x5E2F1D + +static void sip_thread_start(void) +{ + pj_thread_desc *desc; + pj_thread_t *thread; + uint32_t *servant_id; + + servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id)); + if (!servant_id) { + ast_log(LOG_ERROR, "Could not set SIP servant ID in thread-local storage.\n"); + return; + } + *servant_id = SIP_SERVANT_ID; + + desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc)); + if (!desc) { + ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n"); + return; + } + pj_bzero(*desc, sizeof(*desc)); + + if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) { + ast_log(LOG_ERROR, "Couldn't register thread with PJLIB.\n"); + } +} + +int ast_sip_thread_is_servant(void) +{ + uint32_t *servant_id; + + servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id)); + if (!servant_id) { + return 0; + } + + return *servant_id == SIP_SERVANT_ID; +} + +static void remove_request_headers(pjsip_endpoint *endpt) +{ + const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt); + pjsip_hdr *iter = request_headers->next; + + while (iter != request_headers) { + pjsip_hdr *to_erase = iter; + iter = iter->next; + pj_list_erase(to_erase); + } +} + +static int load_module(void) +{ + /* The third parameter is just copied from + * example code from PJLIB. This can be adjusted + * if necessary. + */ + pj_status_t status; + + /* XXX For the time being, create hard-coded threadpool + * options. Just bump up by five threads every time we + * don't have any available threads. Idle threads time + * out after a minute. No maximum size + */ + struct ast_threadpool_options options = { + .version = AST_THREADPOOL_OPTIONS_VERSION, + .auto_increment = 5, + .max_size = 0, + .idle_timeout = 60, + .initial_size = 0, + .thread_start = sip_thread_start, + }; + sip_threadpool = ast_threadpool_create("SIP", NULL, &options); + + if (pj_init() != PJ_SUCCESS) { + return AST_MODULE_LOAD_DECLINE; + } + + if (pjlib_util_init() != PJ_SUCCESS) { + pj_shutdown(); + return AST_MODULE_LOAD_DECLINE; + } + + pj_caching_pool_init(&caching_pool, NULL, 1024 * 1024); + if (pjsip_endpt_create(&caching_pool.factory, "SIP", &ast_pjsip_endpoint) != PJ_SUCCESS) { + ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n"); + goto error; + } + + /* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that, + * we need to stop PJSIP from doing it automatically + */ + remove_request_headers(ast_pjsip_endpoint); + + memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL); + if (!memory_pool) { + ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n"); + goto error; + } + + if (ast_sip_initialize_system()) { + ast_log(LOG_ERROR, "Failed to initialize SIP system configuration. Aborting load\n"); + goto error; + } + + pjsip_tsx_layer_init_module(ast_pjsip_endpoint); + pjsip_ua_init_module(ast_pjsip_endpoint, NULL); + + monitor_continue = 1; + status = pj_thread_create(memory_pool, "SIP", (pj_thread_proc *) &monitor_thread_exec, + NULL, PJ_THREAD_DEFAULT_STACK_SIZE * 2, 0, &monitor_thread); + if (status != PJ_SUCCESS) { + ast_log(LOG_ERROR, "Failed to start SIP monitor thread. Aborting load\n"); + goto error; + } + + ast_sip_initialize_global_headers(); + + if (ast_res_pjsip_initialize_configuration()) { + ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n"); + goto error; + } + + if (ast_sip_initialize_distributor()) { + ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n"); + goto error; + } + + if (ast_sip_initialize_outbound_authentication()) { + ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n"); + goto error; + } + + ast_res_pjsip_init_options_handling(0); + + ast_res_pjsip_init_contact_transports(); + +return AST_MODULE_LOAD_SUCCESS; + +error: + ast_sip_destroy_distributor(); + ast_res_pjsip_destroy_configuration(); + ast_sip_destroy_global_headers(); + if (monitor_thread) { + stop_monitor_thread(); + } + if (memory_pool) { + pj_pool_release(memory_pool); + memory_pool = NULL; + } + if (ast_pjsip_endpoint) { + pjsip_endpt_destroy(ast_pjsip_endpoint); + ast_pjsip_endpoint = NULL; + } + pj_caching_pool_destroy(&caching_pool); + return AST_MODULE_LOAD_DECLINE; +} + +static int reload_module(void) +{ + if (ast_res_pjsip_reload_configuration()) { + return AST_MODULE_LOAD_DECLINE; + } + ast_res_pjsip_init_options_handling(1); + return 0; +} + +static int unload_pjsip(void *data) +{ + if (memory_pool) { + pj_pool_release(memory_pool); + memory_pool = NULL; + } + if (ast_pjsip_endpoint) { + pjsip_endpt_destroy(ast_pjsip_endpoint); + ast_pjsip_endpoint = NULL; + } + pj_caching_pool_destroy(&caching_pool); + return 0; +} + +static int unload_module(void) +{ + ast_sip_destroy_distributor(); + ast_res_pjsip_destroy_configuration(); + ast_sip_destroy_global_headers(); + if (monitor_thread) { + stop_monitor_thread(); + } + /* The thread this is called from cannot call PJSIP/PJLIB functions, + * so we have to push the work to the threadpool to handle + */ + ast_sip_push_task_synchronous(NULL, unload_pjsip, NULL); + + ast_threadpool_shutdown(sip_threadpool); + + return 0; +} + +AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Basic SIP resource", + .load = load_module, + .unload = unload_module, + .reload = reload_module, + .load_pri = AST_MODPRI_CHANNEL_DEPEND - 5, +); |