res_pjsip.c

/*
 * 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 &lt;Number&gt;</literal>,
						or only <literal>&lt;Number&gt;</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="dtmf_mode" 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="media_address">
					<synopsis>IP address used in SDP for media handling</synopsis>
					<description><para>
						At the time of SDP creation, the IP address defined here will be used as
						the media address for individual streams in the SDP.
					</para>
					<note><para>
						Be aware that the <literal>external_media_address</literal> option, set in Transport
						configuration, can also affect the final media address used in the SDP.
					</para></note>
					</description>
				</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>
						An endpoint can be identified in multiple ways. Currently, the only supported
						option is <literal>username</literal>, which matches the endpoint based on the
						username in the From header.
						</para>
						<note><para>Endpoints can also be identified by IP address; however, that method
						of identification is not handled by this configuration option. See the documentation
						for the <literal>identify</literal> configuration section for more details on that
						method of endpoint identification. If this option is set to <literal>username</literal>
						and an <literal>identify</literal> configuration section exists for the endpoint, then
						the endpoint can be identified in multiple ways.</para></note>
						<enumlist>
							<enum name="username" />
						</enumlist>
					</description>
				</configOption>
				<configOption name="redirect_method">
					<synopsis>How redirects received from an endpoint are handled</synopsis>
					<description><para>
						When a redirect is received from an endpoint there are multiple ways it can be handled.
						If this option is set to <literal>user</literal> the user portion of the redirect target
						is treated as an extension within the dialplan and dialed using a Local channel. If this option
						is set to <literal>uri_core</literal> the target URI is returned to the dialing application
						which dials it using the PJSIP channel driver and endpoint originally used. If this option is
						set to <literal>uri_pjsip</literal> the redirect occurs within chan_pjsip itself and is not exposed
						to the core at all. The <literal>uri_pjsip</literal> option has the benefit of being more efficient
						and also supporting multiple potential redirect targets. The con is that since redirection occurs
						within chan_pjsip redirecting information is not forwarded and redirection can not be
						prevented.
						</para>
						<enumlist>
							<enum name="user" />
							<enum name="uri_core" />
							<enum name="uri_pjsip" />
						</enumlist>
					</description>
				</configOption>
				<configOption name="mailboxes">
					<synopsis>Mailbox(es) to be associated with</synopsis>
				</configOption>
				<configOption name="moh_suggest" 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, a full SIP URI must be provided</synopsis>
				</configOption>
				<configOption name="rewrite_contact">
					<synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
					<description><para>
						On inbound SIP messages from this endpoint, the Contact header will be changed to have the
						source IP address and port. This option does not affect outbound messages send to this
						endpoint.
					</para></description>
				</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_diversion" default="yes">
					<synopsis>Send the Diversion header, conveying the diversion
					information to the called user agent</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>
						<warning><para>Transport configuration is not affected by reloads. In order to
						change transports, a full Asterisk restart is required</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="call_group">
					<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="pickup_group">
					<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="named_call_group">
					<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="named_pickup_group">
					<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="device_state_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="t38_udptl" 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="t38_udptl_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="t38_udptl_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="fax_detect" 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="t38_udptl_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="t38_udptl_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="tone_zone">
					<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="record_on_feature" 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="record_off_feature" 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="rtp_engine" default="asterisk">
					<synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis>
				</configOption>
				<configOption name="allow_transfer" default="yes">
					<synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis>
				</configOption>
				<configOption name="sdp_owner" default="-">
					<synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
				</configOption>
				<configOption name="sdp_session" 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="allow_subscribe" default="yes">
					<synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis>
				</configOption>
				<configOption name="sub_min_expiry" default="60">
					<synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis>
				</configOption>
				<configOption name="from_user">
					<synopsis>Username to use in From header for requests to this endpoint.</synopsis>
				</configOption>
				<configOption name="mwi_from_user">
					<synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis>
				</configOption>
				<configOption name="from_domain">
					<synopsis>Domain to user in From header for requests to this endpoint.</synopsis>
				</configOption>
				<configOption name="dtls_verify">
					<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="dtls_rekey">
					<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="dtls_cert_file">
					<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="dtls_private_key">
					<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="dtls_cipher">
					<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="dtls_ca_file">
					<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="dtls_ca_path">
					<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="dtls_setup">
					<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>
				<configOption name="set_var">
					<synopsis>Variable set on a channel involving the endpoint.</synopsis>
					<description><para>
					        When a new channel is created using the endpoint set the specified
						variable(s) on that channel. For multiple channel variables specify
						multiple 'set_var'(s).
					</para></description>
				</configOption>
			</configObject>
			<configObject name="auth">
				<synopsis>Authentication type</synopsis>
				<description><para>
					Authentication objects hold the authentication information for use
					by other objects such as <literal>endpoints</literal> or <literal>registrations</literal>.
					This also allows for multiple objects to use a single auth object. See
					the <literal>auth_type</literal> config option for password style choices.
				</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
						when trying to authenticate an endpoint inbound request. If set to <literal>userpass</literal>
						then we'll read from the 'password' option. For <literal>md5</literal> we'll read
						from 'md5_cred'.
						</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="domain_alias">
				<synopsis>Domain Alias</synopsis>
				<description><para>
					Signifies that a domain is an alias. If the domain on a session is
					not found to match an AoR then this object is used to see if we have
					an alias for the AoR to which the endpoint is binding. This objects
					name as defined in configuration should be the domain alias and a
					config option is provided to specify the domain to be aliased.
				</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>
					<note><para>Changes to transport configuration in pjsip.conf will only be
						effected on a complete restart of Asterisk. A module reload
						will not suffice.</para></note>
				</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 IP address to use in RTP handling</synopsis>
					<description><para>
						When a request or response is sent out, if the destination of the
						message is outside the IP network defined in the option <literal>localnet</literal>,
						and the media address in the SDP is within the localnet network, then the
						media address in the SDP will be rewritten to the value defined for
						<literal>external_media_address</literal>.
					</para></description>
				</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="local_net">
					<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="priv_key_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" />
							<enum name="ws" />
							<enum name="wss" />
						</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>
				<configOption name="tos" default="false">
					<synopsis>Enable TOS for the signalling sent over this transport</synopsis>
					<description>
					<para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
					for more information on this parameter.</para>
					<note><para>This option does not apply to the <replaceable>ws</replaceable>
					or the <replaceable>wss</replaceable> protocols.</para></note>
					</description>
				</configOption>
				<configOption name="cos" default="false">
					<synopsis>Enable COS for the signalling sent over this transport</synopsis>
					<description>
					<para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
					for more information on this parameter.</para>
					<note><para>This option does not apply to the <replaceable>ws</replaceable>
					or the <replaceable>wss</replaceable> protocols.</para></note>
					</description>
				</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>
				<configOption name="outbound_proxy">
					<synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
					<description><para>
						If set the provided URI will be used as the outbound proxy when an
						OPTIONS request is sent to a contact for qualify purposes.
					</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, such as inbound
					registration.
					</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><para>
					Registrations: For Asterisk to match an inbound registration to an endpoint,
					the AoR object name must match the user portion of the SIP URI in the "To:"
					header of the inbound SIP registration. That will usually be equivalent
					to the "user name" set in your hard or soft phones configuration.
				</para></description>
				<configOption name="contact">
					<synopsis>Permanent contacts assigned to AoR</synopsis>
					<description><para>
						Contacts specified will be called whenever referenced
						by <literal>chan_pjsip</literal>.
						</para><para>
						Use a separate "contact=" entry for each contact required. Contacts
						are specified using a SIP URI.
					</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 will be subscribed to.
					More than one mailbox can be specified with a comma-delimited string.</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. This value does
						not affect the number of contacts that can be added with the "contact" option.
						It only limits contacts added through external interaction, such as
						registration.
						</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>
				<configOption name="outbound_proxy">
					<synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
					<description><para>
						If set the provided URI will be used as the outbound proxy when an
						OPTIONS request is sent to a contact for qualify purposes.
					</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="timer_t1" default="500">
					<synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
					<description><para>