Merge the pimp_my_sip branch into trunk.

The pimp_my_sip branch is being merged at this point because
it offers basic functionality, and from an API standpoint, things
are complete.

SIP work is *not* feature-complete; however, with the completion
of the SUBSCRIBE/NOTIFY API, all APIs (except a PUBLISH API) have
been created, and thus it is possible for developers to attempt
to create new SIP work.

API documentation can be found in the doxygen in the code, but
usability documentation is still lacking.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386540 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 files changed, 1966 insertions, 37 deletions

diff --git a/include/asterisk/autoconfig.h.in b/include/asterisk/autoconfig.h.in
index f7294b36e..60f2068e5 100644
--- a/include/asterisk/autoconfig.h.in
+++ b/include/asterisk/autoconfig.h.in
@@ -294,7 +294,7 @@
 /* Define if your system has the GLOB_NOMAGIC headers. */
 #undef HAVE_GLOB_NOMAGIC
 
-/* Define if your system has the GMIME libraries. */
+/* Define to 1 if you have the GMime library. */
 #undef HAVE_GMIME
 
 /* Define to indicate the GSM library */
@@ -306,7 +306,7 @@
 /* Define to indicate that gsm.h has no prefix for its location */
 #undef HAVE_GSM_HEADER
 
-/* Define if your system has the GTK2 libraries. */
+/* Define to 1 if you have the gtk2 library. */
 #undef HAVE_GTK2
 
 /* Define to 1 if you have the Hoard Memory Allocator library. */
@@ -324,7 +324,7 @@
 /* Define to 1 if you have the Iksemel Jabber library. */
 #undef HAVE_IKSEMEL
 
-/* Define if your system has the ILBC libraries. */
+/* Define to 1 if you have the System iLBC library. */
 #undef HAVE_ILBC
 
 /* Define if your system has the UW IMAP Toolkit c-client library. */
@@ -376,7 +376,7 @@
 /* Define to 1 if you have the OpenLDAP library. */
 #undef HAVE_LDAP
 
-/* Define if your system has the LIBEDIT libraries. */
+/* Define to 1 if you have the NetBSD Editline library library. */
 #undef HAVE_LIBEDIT
 
 /* Define to 1 if you have the <libintl.h> header file. */
@@ -551,7 +551,7 @@
 /* Define to indicate presence of the pg_encoding_to_char API. */
 #undef HAVE_PGSQL_pg_encoding_to_char
 
-/* Define if your system has the PJPROJECT libraries. */
+/* Define to 1 if you have the PJPROJECT library. */
 #undef HAVE_PJPROJECT
 
 /* Define to 1 if your system defines IP_PKTINFO. */
@@ -854,19 +854,19 @@
 /* Define to 1 if you have the `strtoq' function. */
 #undef HAVE_STRTOQ
 
-/* Define to 1 if `ifr_ifru.ifru_hwaddr' is a member of `struct ifreq'. */
+/* Define to 1 if `ifr_ifru.ifru_hwaddr' is member of `struct ifreq'. */
 #undef HAVE_STRUCT_IFREQ_IFR_IFRU_IFRU_HWADDR
 
-/* Define to 1 if `uid' is a member of `struct sockpeercred'. */
+/* Define to 1 if `uid' is member of `struct sockpeercred'. */
 #undef HAVE_STRUCT_SOCKPEERCRED_UID
 
-/* Define to 1 if `st_blksize' is a member of `struct stat'. */
+/* Define to 1 if `st_blksize' is member of `struct stat'. */
 #undef HAVE_STRUCT_STAT_ST_BLKSIZE
 
-/* Define to 1 if `cr_uid' is a member of `struct ucred'. */
+/* Define to 1 if `cr_uid' is member of `struct ucred'. */
 #undef HAVE_STRUCT_UCRED_CR_UID
 
-/* Define to 1 if `uid' is a member of `struct ucred'. */
+/* Define to 1 if `uid' is member of `struct ucred'. */
 #undef HAVE_STRUCT_UCRED_UID
 
 /* Define to 1 if you have the mISDN Supplemental Services library. */
@@ -1144,12 +1144,12 @@
 /* Define to the one symbol short name of this package. */
 #undef PACKAGE_TARNAME
 
-/* Define to the home page for this package. */
-#undef PACKAGE_URL
-
 /* Define to the version of this package. */
 #undef PACKAGE_VERSION
 
+/* Define to 1 if the C compiler supports function prototypes. */
+#undef PROTOTYPES
+
 /* Define to necessary symbol if this constant uses a non-standard name on
    your system. */
 #undef PTHREAD_CREATE_JOINABLE
@@ -1169,6 +1169,11 @@
 /* Define to the type of arg 5 for `select'. */
 #undef SELECT_TYPE_ARG5
 
+/* Define to 1 if the `setvbuf' function takes the buffering type as its
+   second argument and the buffer pointer as the third, as on System V before
+   release 3. */
+#undef SETVBUF_REVERSED
+
 /* The size of `char *', as computed by sizeof. */
 #undef SIZEOF_CHAR_P
 
@@ -1204,39 +1209,24 @@
 /* Define to a type of the same size as fd_set.fds_bits[[0]] */
 #undef TYPEOF_FD_SET_FDS_BITS
 
-/* Enable extensions on AIX 3, Interix.  */
+/* Define to 1 if on AIX 3.
+   System headers sometimes define this.
+   We just want to avoid a redefinition error message.  */
 #ifndef _ALL_SOURCE
 # undef _ALL_SOURCE
 #endif
-/* Enable GNU extensions on systems that have them.  */
-#ifndef _GNU_SOURCE
-# undef _GNU_SOURCE
-#endif
-/* Enable threading extensions on Solaris.  */
-#ifndef _POSIX_PTHREAD_SEMANTICS
-# undef _POSIX_PTHREAD_SEMANTICS
-#endif
-/* Enable extensions on HP NonStop.  */
-#ifndef _TANDEM_SOURCE
-# undef _TANDEM_SOURCE
-#endif
-/* Enable general extensions on Solaris.  */
-#ifndef __EXTENSIONS__
-# undef __EXTENSIONS__
-#endif
-
 
 /* Define to 1 if running on Darwin. */
 #undef _DARWIN_UNLIMITED_SELECT
 
-/* Enable large inode numbers on Mac OS X 10.5.  */
-#ifndef _DARWIN_USE_64_BIT_INODE
-# define _DARWIN_USE_64_BIT_INODE 1
-#endif
-
 /* Number of bits in a file offset, on hosts where this is settable. */
 #undef _FILE_OFFSET_BITS
 
+/* Enable GNU extensions on systems that have them.  */
+#ifndef _GNU_SOURCE
+# undef _GNU_SOURCE
+#endif
+
 /* Define to 1 to make fseeko visible on some hosts (e.g. glibc 2.2). */
 #undef _LARGEFILE_SOURCE
 
@@ -1253,6 +1243,20 @@
 /* Define to 1 if you need to in order for `stat' and other things to work. */
 #undef _POSIX_SOURCE
 
+/* Enable extensions on Solaris.  */
+#ifndef __EXTENSIONS__
+# undef __EXTENSIONS__
+#endif
+#ifndef _POSIX_PTHREAD_SEMANTICS
+# undef _POSIX_PTHREAD_SEMANTICS
+#endif
+#ifndef _TANDEM_SOURCE
+# undef _TANDEM_SOURCE
+#endif
+
+/* Define like PROTOTYPES; this can be used by system headers. */
+#undef __PROTOTYPES
+
 /* Define to empty if `const' does not conform to ANSI C. */
 #undef const
 
diff --git a/include/asterisk/res_sip.h b/include/asterisk/res_sip.h
new file mode 100644
index 000000000..7cfc38260
--- /dev/null
+++ b/include/asterisk/res_sip.h
@@ -0,0 +1,1092 @@
+/*
+ * 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.
+ */
+
+#ifndef _RES_SIP_H
+#define _RES_SIP_H
+
+#include "asterisk/stringfields.h"
+/* Needed for struct ast_sockaddr */
+#include "asterisk/netsock2.h"
+/* Needed for linked list macros */
+#include "asterisk/linkedlists.h"
+/* Needed for ast_party_id */
+#include "asterisk/channel.h"
+/* Needed for ast_sorcery */
+#include "asterisk/sorcery.h"
+/* Needed for ast_dnsmgr */
+#include "asterisk/dnsmgr.h"
+/* Needed for pj_sockaddr */
+#include <pjlib.h>
+
+/* Forward declarations of PJSIP stuff */
+struct pjsip_rx_data;
+struct pjsip_module;
+struct pjsip_tx_data;
+struct pjsip_dialog;
+struct pjsip_transport;
+struct pjsip_tpfactory;
+struct pjsip_tls_setting;
+struct pjsip_tpselector;
+
+/*!
+ * \brief Structure for SIP transport information
+ */
+struct ast_sip_transport_state {
+	/*! \brief Transport itself */
+	struct pjsip_transport *transport;
+
+	/*! \brief Transport factory */
+	struct pjsip_tpfactory *factory;
+};
+
+#define SIP_SORCERY_DOMAIN_ALIAS_TYPE "domain_alias"
+
+/*!
+ * Details about a SIP domain alias
+ */
+struct ast_sip_domain_alias {
+	/*! Sorcery object details */
+	SORCERY_OBJECT(details);
+	AST_DECLARE_STRING_FIELDS(
+		/*! Domain to be aliased to */
+		AST_STRING_FIELD(domain);
+	);
+};
+
+/*!
+ * \brief Types of supported transports
+ */
+enum ast_sip_transport_type {
+	AST_SIP_TRANSPORT_UDP,
+	AST_SIP_TRANSPORT_TCP,
+	AST_SIP_TRANSPORT_TLS,
+	/* XXX Websocket ? */
+};
+
+/*! \brief Maximum number of ciphers supported for a TLS transport */
+#define SIP_TLS_MAX_CIPHERS 64
+
+/*
+ * \brief Transport to bind to
+ */
+struct ast_sip_transport {
+	/*! Sorcery object details */
+	SORCERY_OBJECT(details);
+	AST_DECLARE_STRING_FIELDS(
+		/*! Certificate of authority list file */
+		AST_STRING_FIELD(ca_list_file);
+		/*! Public certificate file */
+		AST_STRING_FIELD(cert_file);
+		/*! Optional private key of the certificate file */
+		AST_STRING_FIELD(privkey_file);
+		/*! Password to open the private key */
+		AST_STRING_FIELD(password);
+		/*! External signaling address */
+		AST_STRING_FIELD(external_signaling_address);
+		/*! External media address */
+		AST_STRING_FIELD(external_media_address);
+		/*! Optional domain to use for messages if provided could not be found */
+		AST_STRING_FIELD(domain);
+		);
+	/*! Type of transport */
+	enum ast_sip_transport_type type;
+	/*! Address and port to bind to */
+	pj_sockaddr host;
+	/*! Number of simultaneous asynchronous operations */
+	unsigned int async_operations;
+	/*! Optional external port for signaling */
+	unsigned int external_signaling_port;
+	/*! TLS settings */
+	pjsip_tls_setting tls;
+	/*! Configured TLS ciphers */
+	pj_ssl_cipher ciphers[SIP_TLS_MAX_CIPHERS];
+	/*! Optional local network information, used for NAT purposes */
+	struct ast_ha *localnet;
+	/*! DNS manager for refreshing the external address */
+	struct ast_dnsmgr_entry *external_address_refresher;
+	/*! Optional external address information */
+	struct ast_sockaddr external_address;
+	/*! Transport state information */
+	struct ast_sip_transport_state *state;
+};
+
+/*!
+ * \brief Structure for SIP nat hook information
+ */
+struct ast_sip_nat_hook {
+	/*! Sorcery object details */
+	SORCERY_OBJECT(details);
+	/*! Callback for when a message is going outside of our local network */
+	void (*outgoing_external_message)(struct pjsip_tx_data *tdata, struct ast_sip_transport *transport);
+};
+
+/*!
+ * \brief Contact associated with an address of record
+ */
+struct ast_sip_contact {
+	/*! Sorcery object details, the id is the aor name plus a random string */
+	SORCERY_OBJECT(details);
+	AST_DECLARE_STRING_FIELDS(
+		/*! Full URI of the contact */
+		AST_STRING_FIELD(uri);
+	);
+	/*! Absolute time that this contact is no longer valid after */
+	struct timeval expiration_time;
+};
+
+/*!
+ * \brief A SIP address of record
+ */
+struct ast_sip_aor {
+	/*! Sorcery object details, the id is the AOR name */
+	SORCERY_OBJECT(details);
+	AST_DECLARE_STRING_FIELDS(
+		/*! Voicemail boxes for this AOR */
+		AST_STRING_FIELD(mailboxes);
+	);
+	/*! Minimum expiration time */
+	unsigned int minimum_expiration;
+	/*! Maximum expiration time */
+	unsigned int maximum_expiration;
+	/*! Default contact expiration if one is not provided in the contact */
+	unsigned int default_expiration;
+	/*! Maximum number of external contacts, 0 to disable */
+	unsigned int max_contacts;
+	/*! Whether to remove any existing contacts not related to an incoming REGISTER when it comes in */
+	unsigned int remove_existing;
+	/*! Any permanent configured contacts */
+	struct ao2_container *permanent_contacts;
+};
+
+/*!
+ * \brief DTMF modes for SIP endpoints
+ */
+enum ast_sip_dtmf_mode {
+	/*! No DTMF to be used */
+	AST_SIP_DTMF_NONE,
+	/* XXX Should this be 2833 instead? */
+	/*! Use RFC 4733 events for DTMF */
+	AST_SIP_DTMF_RFC_4733,
+	/*! Use DTMF in the audio stream */
+	AST_SIP_DTMF_INBAND,
+	/*! Use SIP INFO DTMF (blech) */
+	AST_SIP_DTMF_INFO,
+};
+
+/*!
+ * \brief Methods of storing SIP digest authentication credentials.
+ *
+ * Note that both methods result in MD5 digest authentication being
+ * used. The two methods simply alter how Asterisk determines the
+ * credentials for a SIP authentication
+ */
+enum ast_sip_auth_type {
+	/*! Credentials stored as a username and password combination */
+	AST_SIP_AUTH_TYPE_USER_PASS,
+	/*! Credentials stored as an MD5 sum */
+	AST_SIP_AUTH_TYPE_MD5,
+};
+
+#define SIP_SORCERY_AUTH_TYPE "auth"
+
+struct ast_sip_auth {
+	/* Sorcery ID of the auth is its name */
+	SORCERY_OBJECT(details);
+	AST_DECLARE_STRING_FIELDS(
+		/* Identification for these credentials */
+		AST_STRING_FIELD(realm);
+		/* Authentication username */
+		AST_STRING_FIELD(auth_user);
+		/* Authentication password */
+		AST_STRING_FIELD(auth_pass);
+		/* Authentication credentials in MD5 format (hash of user:realm:pass) */
+		AST_STRING_FIELD(md5_creds);
+	);
+	/* The time period (in seconds) that a nonce may be reused */
+	unsigned int nonce_lifetime;
+	/* Used to determine what to use when authenticating */
+	enum ast_sip_auth_type type;
+};
+
+/*!
+ * \brief Different methods by which incoming requests can be matched to endpoints
+ */
+enum ast_sip_endpoint_identifier_type {
+	/*! Identify based on user name in From header */
+	AST_SIP_ENDPOINT_IDENTIFY_BY_USERNAME = (1 << 0),
+	/*! Identify based on source location of the SIP message */
+	AST_SIP_ENDPOINT_IDENTIFY_BY_LOCATION = (1 << 1),
+};
+
+enum ast_sip_session_refresh_method {
+	/*! Use reinvite to negotiate direct media */
+	AST_SIP_SESSION_REFRESH_METHOD_INVITE,
+	/*! Use UPDATE to negotiate direct media */
+	AST_SIP_SESSION_REFRESH_METHOD_UPDATE,
+};
+
+enum ast_sip_direct_media_glare_mitigation {
+	/*! Take no special action to mitigate reinvite glare */
+	AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE,
+	/*! Do not send an initial direct media session refresh on outgoing call legs
+	 * Subsequent session refreshes will be sent no matter the session direction
+	 */
+	AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_OUTGOING,
+	/*! Do not send an initial direct media session refresh on incoming call legs
+	 * Subsequent session refreshes will be sent no matter the session direction
+	 */
+	AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_INCOMING,
+};
+
+/*!
+ * \brief An entity with which Asterisk communicates
+ */
+struct ast_sip_endpoint {
+	SORCERY_OBJECT(details);
+	AST_DECLARE_STRING_FIELDS(
+		/*! Context to send incoming calls to */
+		AST_STRING_FIELD(context);
+		/*! Name of an explicit transport to use */
+		AST_STRING_FIELD(transport);
+		/*! Outbound proxy to use */
+		AST_STRING_FIELD(outbound_proxy);
+		/*! Explicit AORs to dial if none are specified */
+		AST_STRING_FIELD(aors);
+                /*! Musiconhold class to suggest that the other side use when placing on hold */
+                AST_STRING_FIELD(mohsuggest);
+		/*! Optional external media address to use in SDP */
+		AST_STRING_FIELD(external_media_address);
+		/*! Configured voicemail boxes for this endpoint. Used for MWI */
+		AST_STRING_FIELD(mailboxes);
+	);
+	/*! Identification information for this endpoint */
+	struct ast_party_id id;
+	/*! Domain to which this endpoint belongs */
+	struct ast_sip_domain *domain;
+	/*! Address of record for incoming registrations */
+	struct ast_sip_aor *aor;
+	/*! Codec preferences */
+	struct ast_codec_pref prefs;
+	/*! Configured codecs */
+	struct ast_format_cap *codecs;
+	/*! Names of inbound authentication credentials */
+	const char **sip_inbound_auths;
+	/*! Number of configured auths */
+	size_t num_inbound_auths;
+	/*! Names of outbound authentication credentials */
+	const char **sip_outbound_auths;
+	/*! Number of configured outbound auths */
+	size_t num_outbound_auths;
+	/*! DTMF mode to use with this endpoint */
+	enum ast_sip_dtmf_mode dtmf;
+	/*! Whether IPv6 RTP is enabled or not */
+	unsigned int rtp_ipv6;
+	/*! Whether symmetric RTP is enabled or not */
+	unsigned int rtp_symmetric;
+	/*! Whether ICE support is enabled or not */
+	unsigned int ice_support;
+	/*! Whether to use the "ptime" attribute received from the endpoint or not */
+	unsigned int use_ptime;
+	/*! Whether to force using the source IP address/port for sending responses */
+	unsigned int force_rport;
+	/*! Whether to rewrite the Contact header with the source IP address/port or not */
+	unsigned int rewrite_contact;
+	/*! Enabled SIP extensions */
+	unsigned int extensions;
+	/*! Minimum session expiration period, in seconds */
+	unsigned int min_se;
+	/*! Session expiration period, in seconds */
+	unsigned int sess_expires;
+	/*! List of outbound registrations */
+	AST_LIST_HEAD_NOLOCK(, ast_sip_registration) registrations;
+	/*! Frequency to send OPTIONS requests to endpoint. 0 is disabled. */
+	unsigned int qualify_frequency;
+	/*! Method(s) by which the endpoint should be identified. */
+	enum ast_sip_endpoint_identifier_type ident_method;
+	/*! Boolean indicating if direct_media is permissible */
+	unsigned int direct_media;
+	/*! When using direct media, which method should be used */
+	enum ast_sip_session_refresh_method direct_media_method;
+	/*! Take steps to mitigate glare for direct media */
+	enum ast_sip_direct_media_glare_mitigation direct_media_glare_mitigation;
+	/*! Do not attempt direct media session refreshes if a media NAT is detected */
+	unsigned int disable_direct_media_on_nat;
+	/*! Do we trust the endpoint with our outbound identity? */
+	unsigned int trust_id_outbound;
+	/*! Do we trust identity information that originates externally (e.g. P-Asserted-Identity header)? */
+	unsigned int trust_id_inbound;
+	/*! Do we send P-Asserted-Identity headers to this endpoint? */
+	unsigned int send_pai;
+	/*! Do we send Remote-Party-ID headers to this endpoint? */
+	unsigned int send_rpid;
+	/*! Should unsolicited MWI be aggregated into a single NOTIFY? */
+	unsigned int aggregate_mwi;
+};
+
+/*!
+ * \brief Possible returns from ast_sip_check_authentication
+ */
+enum ast_sip_check_auth_result {
+    /*! Authentication needs to be challenged */
+    AST_SIP_AUTHENTICATION_CHALLENGE,
+    /*! Authentication succeeded */
+    AST_SIP_AUTHENTICATION_SUCCESS,
+    /*! Authentication failed */
+    AST_SIP_AUTHENTICATION_FAILED,
+    /*! Authentication encountered some internal error */
+    AST_SIP_AUTHENTICATION_ERROR,
+};
+
+/*!
+ * \brief An interchangeable way of handling digest authentication for SIP.
+ *
+ * An authenticator is responsible for filling in the callbacks provided below. Each is called from a publicly available
+ * function in res_sip. The authenticator can use configuration or other local policy to determine whether authentication
+ * should take place and what credentials should be used when challenging and authenticating a request.
+ */
+struct ast_sip_authenticator {
+    /*!
+     * \brief Check if a request requires authentication
+     * See ast_sip_requires_authentication for more details
+     */
+    int (*requires_authentication)(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata);
+	/*!
+	 * \brief Check that an incoming request passes authentication.
+	 *
+	 * The tdata parameter is useful for adding information such as digest challenges.
+	 *
+	 * \param endpoint The endpoint sending the incoming request
+	 * \param rdata The incoming request
+	 * \param tdata Tentative outgoing request.
+	 */
+	enum ast_sip_check_auth_result (*check_authentication)(struct ast_sip_endpoint *endpoint,
+			pjsip_rx_data *rdata, pjsip_tx_data *tdata);
+};
+
+/*!
+ * \brief an interchangeable way of responding to authentication challenges
+ *
+ * An outbound authenticator takes incoming challenges and formulates a new SIP request with
+ * credentials.
+ */
+struct ast_sip_outbound_authenticator {
+	/*!
+	 * \brief Create a new request with authentication credentials
+	 *
+	 * \param auths An array of IDs of auth sorcery objects
+	 * \param num_auths The number of IDs in the array
+	 * \param challenge The SIP response with authentication challenge(s)
+	 * \param tsx The transaction in which the challenge was received
+	 * \param new_request The new SIP request with challenge response(s)
+	 * \retval 0 Successfully created new request
+	 * \retval -1 Failed to create a new request
+	 */
+	int (*create_request_with_auth)(const char **auths, size_t num_auths, struct pjsip_rx_data *challenge,
+			struct pjsip_transaction *tsx, struct pjsip_tx_data **new_request);
+};
+
+/*!
+ * \brief An entity responsible for identifying the source of a SIP message
+ */
+struct ast_sip_endpoint_identifier {
+    /*!
+     * \brief Callback used to identify the source of a message.
+     * See ast_sip_identify_endpoint for more details
+     */
+    struct ast_sip_endpoint *(*identify_endpoint)(pjsip_rx_data *rdata);
+};
+
+/*!
+ * \brief Register a SIP service in Asterisk.
+ *
+ * This is more-or-less a wrapper around pjsip_endpt_register_module().
+ * Registering a service makes it so that PJSIP will call into the
+ * service at appropriate times. For more information about PJSIP module
+ * callbacks, see the PJSIP documentation. Asterisk modules that call
+ * this function will likely do so at module load time.
+ *
+ * \param module The module that is to be registered with PJSIP
+ * \retval 0 Success
+ * \retval -1 Failure
+ */
+int ast_sip_register_service(pjsip_module *module);
+
+/*!
+ * This is the opposite of ast_sip_register_service().  Unregistering a
+ * service means that PJSIP will no longer call into the module any more.
+ * This will likely occur when an Asterisk module is unloaded.
+ *
+ * \param module The PJSIP module to unregister
+ */
+void ast_sip_unregister_service(pjsip_module *module);
+
+/*!
+ * \brief Register a SIP authenticator
+ *
+ * An authenticator has three main purposes:
+ * 1) Determining if authentication should be performed on an incoming request
+ * 2) Gathering credentials necessary for issuing an authentication challenge
+ * 3) Authenticating a request that has credentials
+ *
+ * Asterisk provides a default authenticator, but it may be replaced by a
+ * custom one if desired.
+ *
+ * \param auth The authenticator to register
+ * \retval 0 Success
+ * \retval -1 Failure
+ */
+int ast_sip_register_authenticator(struct ast_sip_authenticator *auth);
+
+/*!
+ * \brief Unregister a SIP authenticator
+ *
+ * When there is no authenticator registered, requests cannot be challenged
+ * or authenticated.
+ *
+ * \param auth The authenticator to unregister
+ */
+void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth);
+
+ /*!
+ * \brief Register an outbound SIP authenticator
+ *
+ * An outbound authenticator is responsible for creating responses to
+ * authentication challenges by remote endpoints.
+ *
+ * \param auth The authenticator to register
+ * \retval 0 Success
+ * \retval -1 Failure
+ */
+int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *outbound_auth);
+
+/*!
+ * \brief Unregister an outbound SIP authenticator
+ *
+ * When there is no outbound authenticator registered, authentication challenges
+ * will be handled as any other final response would be.
+ *
+ * \param auth The authenticator to unregister
+ */
+void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth);
+
+/*!
+ * \brief Register a SIP endpoint identifier
+ *
+ * An endpoint identifier's purpose is to determine which endpoint a given SIP
+ * message has come from.
+ *
+ * Multiple endpoint identifiers may be registered so that if an endpoint
+ * cannot be identified by one identifier, it may be identified by another.
+ *
+ * Asterisk provides two endpoint identifiers. One identifies endpoints based
+ * on the user part of the From header URI. The other identifies endpoints based
+ * on the source IP address.
+ *
+ * If the order in which endpoint identifiers is run is important to you, then
+ * be sure to load individual endpoint identifier modules in the order you wish
+ * for them to be run in modules.conf
+ *
+ * \param identifier The SIP endpoint identifier to register
+ * \retval 0 Success
+ * \retval -1 Failure
+ */
+int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier);
+
+/*!
+ * \brief Unregister a SIP endpoint identifier
+ *
+ * This stops an endpoint identifier from being used.
+ *
+ * \param identifier The SIP endoint identifier to unregister
+ */
+void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier);
+
+/*!
+ * \brief Allocate a new SIP endpoint
+ *
+ * This will return an endpoint with its refcount increased by one. This reference
+ * can be released using ao2_ref().
+ *
+ * \param name The name of the endpoint.
+ * \retval NULL Endpoint allocation failed
+ * \retval non-NULL The newly allocated endpoint
+ */
+void *ast_sip_endpoint_alloc(const char *name);
+
+/*!
+ * \brief Get a pointer to the PJSIP endpoint.
+ *
+ * This is useful when modules have specific information they need
+ * to register with the PJSIP core.
+ * \retval NULL endpoint has not been created yet.
+ * \retval non-NULL PJSIP endpoint.
+ */
+pjsip_endpoint *ast_sip_get_pjsip_endpoint(void);
+
+/*!
+ * \brief Get a pointer to the SIP sorcery structure.
+ *
+ * \retval NULL sorcery has not been initialized
+ * \retval non-NULL sorcery structure
+ */
+struct ast_sorcery *ast_sip_get_sorcery(void);
+
+/*!
+ * \brief Initialize transport support on a sorcery instance
+ *
+ * \param sorcery The sorcery instance
+ *
+ * \retval -1 failure
+ * \retval 0 success
+ */
+int ast_sip_initialize_sorcery_transport(struct ast_sorcery *sorcery);
+
+/*!
+ * \brief Initialize location support on a sorcery instance
+ *
+ * \param sorcery The sorcery instance
+ *
+ * \retval -1 failure
+ * \retval 0 success
+ */
+int ast_sip_initialize_sorcery_location(struct ast_sorcery *sorcery);
+
+/*!
+ * \brief Retrieve a named AOR
+ *
+ * \param aor_name Name of the AOR
+ *
+ * \retval NULL if not found
+ * \retval non-NULL if found
+ */
+struct ast_sip_aor *ast_sip_location_retrieve_aor(const char *aor_name);
+
+/*!
+ * \brief Retrieve the first bound contact for an AOR
+ *
+ * \param aor Pointer to the AOR
+ * \retval NULL if no contacts available
+ * \retval non-NULL if contacts available
+ */
+struct ast_sip_contact *ast_sip_location_retrieve_first_aor_contact(const struct ast_sip_aor *aor);
+
+/*!
+ * \brief Retrieve all contacts currently available for an AOR
+ *
+ * \param aor Pointer to the AOR
+ *
+ * \retval NULL if no contacts available
+ * \retval non-NULL if contacts available
+ */
+struct ao2_container *ast_sip_location_retrieve_aor_contacts(const struct ast_sip_aor *aor);
+
+/*!
+ * \brief Retrieve the first bound contact from a list of AORs
+ *
+ * \param aor_list A comma-separated list of AOR names
+ * \retval NULL if no contacts available
+ * \retval non-NULL if contacts available
+ */
+struct ast_sip_contact *ast_sip_location_retrieve_contact_from_aor_list(const char *aor_list);
+
+/*!
+ * \brief Retrieve a named contact
+ *
+ * \param contact_name Name of the contact
+ *
+ * \retval NULL if not found
+ * \retval non-NULL if found
+ */
+struct ast_sip_contact *ast_sip_location_retrieve_contact(const char *contact_name);
+
+/*!
+ * \brief Add a new contact to an AOR
+ *
+ * \param aor Pointer to the AOR
+ * \param uri Full contact URI
+ * \param expiration_time Optional expiration time of the contact
+ *
+ * \retval -1 failure
+ * \retval 0 success
+ */
+int ast_sip_location_add_contact(struct ast_sip_aor *aor, const char *uri, struct timeval expiration_time);
+
+/*!
+ * \brief Update a contact
+ *
+ * \param contact New contact object with details
+ *
+ * \retval -1 failure
+ * \retval 0 success
+ */
+int ast_sip_location_update_contact(struct ast_sip_contact *contact);
+
+/*!
+* \brief Delete a contact
+*
+* \param contact Contact object to delete
+*
+* \retval -1 failure
+* \retval 0 success
+*/
+int ast_sip_location_delete_contact(struct ast_sip_contact *contact);
+
+/*!
+ * \brief Initialize domain aliases support on a sorcery instance
+ *
+ * \param sorcery The sorcery instance
+ *
+ * \retval -1 failure
+ * \retval 0 success
+ */
+int ast_sip_initialize_sorcery_domain_alias(struct ast_sorcery *sorcery);
+
+/*!
+ * \brief Initialize authentication support on a sorcery instance
+ *
+ * \param sorcery The sorcery instance
+ *
+ * \retval -1 failure
+ * \retval 0 success
+ */
+int ast_sip_initialize_sorcery_auth(struct ast_sorcery *sorcery);
+
+/*!
+ * \brief Callback called when an outbound request with authentication credentials is to be sent in dialog
+ *
+ * This callback will have the created request on it. The callback's purpose is to do any extra
+ * housekeeping that needs to be done as well as to send the request out.
+ *
+ * This callback is only necessary if working with a PJSIP API that sits between the application
+ * and the dialog layer.
+ *
+ * \param dlg The dialog to which the request belongs
+ * \param tdata The created request to be sent out
+ * \param user_data Data supplied with the callback
+ *
+ * \retval 0 Success
+ * \retval -1 Failure
+ */
+typedef int (*ast_sip_dialog_outbound_auth_cb)(pjsip_dialog *dlg, pjsip_tx_data *tdata, void *user_data);
+
+/*!
+ * \brief Set up outbound authentication on a SIP dialog
+ *
+ * This sets up the infrastructure so that all requests associated with a created dialog
+ * can be re-sent with authentication credentials if the original request is challenged.
+ *
+ * \param dlg The dialog on which requests will be authenticated
+ * \param endpoint The endpoint whom this dialog pertains to
+ * \param cb Callback to call to send requests with authentication
+ * \param user_data Data to be provided to the callback when it is called
+ *
+ * \retval 0 Success
+ * \retval -1 Failure
+ */
+int ast_sip_dialog_setup_outbound_authentication(pjsip_dialog *dlg, const struct ast_sip_endpoint *endpoint,
+		ast_sip_dialog_outbound_auth_cb cb, void *user_data);
+
+/*!
+ * \brief Initialize the distributor module
+ *
+ * The distributor module is responsible for taking an incoming
+ * SIP message and placing it into the threadpool. Once in the threadpool,
+ * the distributor will perform endpoint lookups and authentication, and
+ * then distribute the message up the stack to any further modules.
+ *
+ * \retval -1 Failure
+ * \retval 0 Success
+ */
+int ast_sip_initialize_distributor(void);
+
+/*!
+ * \page Threading model for SIP
+ *
+ * There are three major types of threads that SIP will have to deal with:
+ * \li Asterisk threads
+ * \li PJSIP threads
+ * \li SIP threadpool threads (a.k.a. "servants")
+ *
+ * \par Asterisk Threads
+ *
+ * Asterisk threads are those that originate from outside of SIP but within
+ * Asterisk. The most common of these threads are PBX (channel) threads and
+ * the autoservice thread. Most interaction with these threads will be through
+ * channel technology callbacks. Within these threads, it is fine to handle
+ * Asterisk data from outside of SIP, but any handling of SIP data should be
+ * left to servants, \b especially if you wish to call into PJSIP for anything.
+ * Asterisk threads are not registered with PJLIB, so attempting to call into
+ * PJSIP will cause an assertion to be triggered, thus causing the program to
+ * crash.
+ *
+ * \par PJSIP Threads
+ *
+ * PJSIP threads are those that originate from handling of PJSIP events, such
+ * as an incoming SIP request or response, or a transaction timeout. The role
+ * of these threads is to process information as quickly as possible so that
+ * the next item on the SIP socket(s) can be serviced. On incoming messages,
+ * Asterisk automatically will push the request to a servant thread. When your
+ * module callback is called, processing will already be in a servant. However,
+ * for other PSJIP events, such as transaction state changes due to timer
+ * expirations, your module will be called into from a PJSIP thread. If you
+ * are called into from a PJSIP thread, then you should push whatever processing
+ * is needed to a servant as soon as possible. You can discern if you are currently
+ * in a SIP servant thread using the \ref ast_sip_thread_is_servant function.
+ *
+ * \par Servants
+ *
+ * Servants are where the bulk of SIP work should be performed. These threads
+ * exist in order to do the work that Asterisk threads and PJSIP threads hand
+ * off to them. Servant threads register themselves with PJLIB, meaning that
+ * they are capable of calling PJSIP and PJLIB functions if they wish.
+ *
+ * \par Serializer
+ *
+ * Tasks are handed off to servant threads using the API call \ref ast_sip_push_task.
+ * The first parameter of this call is a serializer. If this pointer
+ * is NULL, then the work will be handed off to whatever servant can currently handle
+ * the task. If this pointer is non-NULL, then the task will not be executed until
+ * previous tasks pushed with the same serializer have completed. For more information
+ * on serializers and the benefits they provide, see \ref ast_threadpool_serializer
+ *
+ * \note
+ *
+ * Do not make assumptions about individual threads based on a corresponding serializer.
+ * In other words, just because several tasks use the same serializer when being pushed
+ * to servants, it does not mean that the same thread is necessarily going to execute those
+ * tasks, even though they are all guaranteed to be executed in sequence.
+ */
+
+/*!
+ * \brief Create a new serializer for SIP tasks
+ *
+ * See \ref ast_threadpool_serializer for more information on serializers.
+ * SIP creates serializers so that tasks operating on similar data will run
+ * in sequence.
+ *
+ * \retval NULL Failure
+ * \retval non-NULL Newly-created serializer
+ */
+struct ast_taskprocessor *ast_sip_create_serializer(void);
+
+/*!
+ * \brief Set a serializer on a SIP dialog so requests and responses are automatically serialized
+ *
+ * Passing a NULL serializer is a way to remove a serializer from a dialog.
+ *
+ * \param dlg The SIP dialog itself
+ * \param serializer The serializer to use
+ */
+void ast_sip_dialog_set_serializer(pjsip_dialog *dlg, struct ast_taskprocessor *serializer);
+
+/*!
+ * \brief Set an endpoint on a SIP dialog so in-dialog requests do not undergo endpoint lookup.
+ *
+ * \param dlg The SIP dialog itself
+ * \param endpoint The endpoint that this dialog is communicating with
+ */
+void ast_sip_dialog_set_endpoint(pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint);
+
+/*!
+ * \brief Get the endpoint associated with this dialog
+ *
+ * This function increases the refcount of the endpoint by one. Release
+ * the reference once you are finished with the endpoint.
+ *
+ * \param dlg The SIP dialog from which to retrieve the endpoint
+ * \retval NULL No endpoint associated with this dialog
+ * \retval non-NULL The endpoint.
+ */
+struct ast_sip_endpoint *ast_sip_dialog_get_endpoint(pjsip_dialog *dlg);
+
+/*!
+ * \brief Pushes a task to SIP servants
+ *
+ * This uses the serializer provided to determine how to push the task.
+ * If the serializer is NULL, then the task will be pushed to the
+ * servants directly. If the serializer is non-NULL, then the task will be
+ * queued behind other tasks associated with the same serializer.
+ *
+ * \param serializer The serializer to which the task belongs. Can be NULL
+ * \param sip_task The task to execute
+ * \param task_data The parameter to pass to the task when it executes
+ * \retval 0 Success
+ * \retval -1 Failure
+ */
+int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data);
+
+/*!
+ * \brief Push a task to SIP servants and wait for it to complete
+ *
+ * Like \ref ast_sip_push_task except that it blocks until the task completes.
+ *
+ * \warning \b Never use this function in a SIP servant thread. This can potentially
+ * cause a deadlock. If you are in a SIP servant thread, just call your function
+ * in-line.
+ *
+ * \param serializer The SIP serializer to which the task belongs. May be NULL.
+ * \param sip_task The task to execute
+ * \param task_data The parameter to pass to the task when it executes
+ * \retval 0 Success
+ * \retval -1 Failure
+ */
+int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data);
+
+/*!
+ * \brief Determine if the current thread is a SIP servant thread
+ *
+ * \retval 0 This is not a SIP servant thread
+ * \retval 1 This is a SIP servant thread
+ */
+int ast_sip_thread_is_servant(void);
+
+/*!
+ * \brief SIP body description
+ *
+ * This contains a type and subtype that will be added as
+ * the "Content-Type" for the message as well as the body
+ * text.
+ */
+struct ast_sip_body {
+	/*! Type of the body, such as "application" */
+	const char *type;
+	/*! Subtype of the body, such as "sdp" */
+	const char *subtype;
+	/*! The text to go in the body */
+	const char *body_text;
+};
+
+/*!
+ * \brief General purpose method for creating a dialog with an endpoint
+ *
+ * \param endpoint A pointer to the endpoint
+ * \param aor_name Optional name of the AOR to target, may even be an explicit SIP URI
+ * \param request_user Optional user to place into the target URI
+ *
+ * \retval non-NULL success
+ * \retval NULL failure
+ */
+ pjsip_dialog *ast_sip_create_dialog(const struct ast_sip_endpoint *endpoint, const char *aor_name, const char *request_user);
+
+/*!
+ * \brief General purpose method for creating a SIP request
+ *
+ * Its typical use would be to create one-off requests such as an out of dialog
+ * SIP MESSAGE.
+ *
+ * The request can either be in- or out-of-dialog. If in-dialog, the
+ * dlg parameter MUST be present. If out-of-dialog the endpoint parameter
+ * MUST be present. If both are present, then we will assume that the message
+ * is to be sent in-dialog.
+ *
+ * The uri parameter can be specified if the request should be sent to an explicit
+ * URI rather than one configured on the endpoint.
+ *
+ * \param method The method of the SIP request to send
+ * \param dlg Optional. If specified, the dialog on which to request the message.
+ * \param endpoint Optional. If specified, the request will be created out-of-dialog
+ * to the endpoint.
+ * \param uri Optional. If specified, the request will be sent to this URI rather
+ * than one configured for the endpoint.
+ * \param[out] tdata The newly-created request
+ * \retval 0 Success
+ * \retval -1 Failure
+ */
+int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
+		struct ast_sip_endpoint *endpoint, const char *uri, pjsip_tx_data **tdata);
+
+/*!
+ * \brief General purpose method for sending a SIP request
+ *
+ * This is a companion function for \ref ast_sip_create_request. The request
+ * created there can be passed to this function, though any request may be
+ * passed in.
+ *
+ * This will automatically set up handling outbound authentication challenges if
+ * they arrive.
+ *
+ * \param tdata The request to send
+ * \param dlg Optional. If specified, the dialog on which the request should be sent
+ * \param endpoint Optional. If specified, the request is sent out-of-dialog to the endpoint.
+ * \retval 0 Success
+ * \retval -1 Failure
+ */
+int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint);
+
+/*!
+ * \brief Determine if an incoming request requires authentication
+ *
+ * This calls into the registered authenticator's requires_authentication callback
+ * in order to determine if the request requires authentication.
+ *
+ * If there is no registered authenticator, then authentication will be assumed
+ * not to be required.
+ *
+ * \param endpoint The endpoint from which the request originates
+ * \param rdata The incoming SIP request
+ * \retval non-zero The request requires authentication
+ * \retval 0 The request does not require authentication
+ */
+int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata);
+
+/*!
+ * \brief Method to determine authentication status of an incoming request
+ *
+ * This will call into a registered authenticator. The registered authenticator will
+ * do what is necessary to determine whether the incoming request passes authentication.
+ * A tentative response is passed into this function so that if, say, a digest authentication
+ * challenge should be sent in the ensuing response, it can be added to the response.
+ *
+ * \param endpoint The endpoint from the request was sent
+ * \param rdata The request to potentially authenticate
+ * \param tdata Tentative response to the request
+ * \return The result of checking authentication.
+ */
+enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
+		pjsip_rx_data *rdata, pjsip_tx_data *tdata);
+
+/*!
+ * \brief Create a response to an authentication challenge
+ *
+ * This will call into an outbound authenticator's create_request_with_auth callback
+ * to create a new request with authentication credentials. See the create_request_with_auth
+ * callback in the \ref ast_sip_outbound_authenticator structure for details about
+ * the parameters and return values.
+ */
+int ast_sip_create_request_with_auth(const char **auths, size_t num_auths, pjsip_rx_data *challenge,
+		pjsip_transaction *tsx, pjsip_tx_data **new_request);
+
+/*!
+ * \brief Determine the endpoint that has sent a SIP message
+ *
+ * This will call into each of the registered endpoint identifiers'
+ * identify_endpoint() callbacks until one returns a non-NULL endpoint.
+ * This will return an ao2 object. Its reference count will need to be
+ * decremented when completed using the endpoint.
+ *
+ * \param rdata The inbound SIP message to use when identifying the endpoint.
+ * \retval NULL No matching endpoint
+ * \retval non-NULL The matching endpoint
+ */
+struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata);
+
+/*!
+ * \brief Add a header to an outbound SIP message
+ *
+ * \param tdata The message to add the header to
+ * \param name The header name
+ * \param value The header value
+ * \retval 0 Success
+ * \retval -1 Failure
+ */
+int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value);
+
+/*!
+ * \brief Add a body to an outbound SIP message
+ *
+ * If this is called multiple times, the latest body will replace the current
+ * body.
+ *
+ * \param tdata The message to add the body to
+ * \param body The message body to add
+ * \retval 0 Success
+ * \retval -1 Failure
+ */
+int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body);
+
+/*!
+ * \brief Add a multipart body to an outbound SIP message
+ *
+ * This will treat each part of the input array as part of a multipart body and
+ * add each part to the SIP message.
+ *
+ * \param tdata The message to add the body to
+ * \param bodies The parts of the body to add
+ * \retval 0 Success
+ * \retval -1 Failure
+ */
+int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies);
+
+/*!
+ * \brief Append body data to a SIP message
+ *
+ * This acts mostly the same as ast_sip_add_body, except that rather than replacing
+ * a body if it currently exists, it appends data to an existing body.
+ *
+ * \param tdata The message to append the body to
+ * \param body The string to append to the end of the current body
+ * \retval 0 Success
+ * \retval -1 Failure
+ */
+int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text);
+
+/*!
+ * \brief Copy a pj_str_t into a standard character buffer.
+ *
+ * pj_str_t is not NULL-terminated. Any place that expects a NULL-
+ * terminated string needs to have the pj_str_t copied into a separate
+ * buffer.
+ *
+ * This method copies the pj_str_t contents into the destination buffer
+ * and NULL-terminates the buffer.
+ *
+ * \param dest The destination buffer
+ * \param src The pj_str_t to copy
+ * \param size The size of the destination buffer.
+ */
+void ast_copy_pj_str(char *dest, pj_str_t *src, size_t size);
+
+/*!
+ * \brief Get the looked-up endpoint on an out-of dialog request or response
+ *
+ * The function may ONLY be called on out-of-dialog requests or responses. For
+ * in-dialog requests and responses, it is required that the user of the dialog
+ * has the looked-up endpoint stored locally.
+ *
+ * This function should never return NULL if the message is out-of-dialog. It will
+ * always return NULL if the message is in-dialog.
+ *
+ * This function will increase the reference count of the returned endpoint by one.
+ * Release your reference using the ao2_ref function when finished.
+ *
+ * \param rdata Out-of-dialog request or response
+ * \return The looked up endpoint
+ */
+struct ast_sip_endpoint *ast_pjsip_rdata_get_endpoint(pjsip_rx_data *rdata);
+
+/*!
+ * \brief Retrieve relevant SIP auth structures from sorcery
+ *
+ * \param auth_names The sorcery IDs of auths to retrieve
+ * \param num_auths The number of auths to retrieve
+ * \param[out] out The retrieved auths are stored here
+ */
+int ast_sip_retrieve_auths(const char *auth_names[], size_t num_auths, struct ast_sip_auth **out);
+
+/*!
+ * \brief Clean up retrieved auth structures from memory
+ *
+ * Call this function once you have completed operating on auths
+ * retrieved from \ref ast_sip_retrieve_auths
+ *
+ * \param auths An array of auth structures to clean up
+ * \param num_auths The number of auths in the array
+ */
+void ast_sip_cleanup_auths(struct ast_sip_auth *auths[], size_t num_auths);
+
+#endif /* _RES_SIP_H */
diff --git a/include/asterisk/res_sip_pubsub.h b/include/asterisk/res_sip_pubsub.h
new file mode 100644
index 000000000..33614b285
--- /dev/null
+++ b/include/asterisk/res_sip_pubsub.h
@@ -0,0 +1,346 @@
+/*
+ * 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.
+ */
+
+#ifndef _RES_SIP_PUBSUB_H
+#define _RES_SIP_PUBSUB_H
+
+#include "asterisk/linkedlists.h"
+
+/* Forward declarations */
+struct pjsip_rx_data;
+struct pjsip_tx_data;
+struct pjsip_evsub;
+struct ast_sip_endpoint;
+struct ast_datastore;
+struct ast_datastore_info;
+
+/*!
+ * \brief Opaque structure representing an RFC 3265 SIP subscription
+ */
+struct ast_sip_subscription;
+ 
+/*!
+ * \brief Role for the subscription that is being created
+ */
+enum ast_sip_subscription_role {
+    /* Sending SUBSCRIBEs, receiving NOTIFYs */
+    AST_SIP_SUBSCRIBER,
+    /* Sending NOTIFYs, receiving SUBSCRIBEs */
+    AST_SIP_NOTIFIER,
+};
+
+/*!
+ * \brief Data for responses to SUBSCRIBEs and NOTIFIEs
+ *
+ * Some of PJSIP's evsub callbacks expect us to provide them
+ * with data so that they can craft a response rather than have
+ * us create our own response.
+ *
+ * Filling in the structure is optional, since the framework
+ * will automatically respond with a 200 OK response if we do
+ * not provide it with any additional data.
+ */
+struct ast_sip_subscription_response_data {
+    /*! Status code of the response */
+    int status_code;
+    /*! Optional status text */
+    const char *status_text;
+    /*! Optional additional headers to add to the response */
+    struct ast_variable *headers;
+    /*! Optional body to add to the response */
+    struct ast_sip_body *body;
+};
+
+#define AST_SIP_MAX_ACCEPT 32
+
+struct ast_sip_subscription_handler {
+    /*! The name of the event this handler deals with */
+    const char *event_name;
+    /*! The types of body this handler accepts */
+    const char *accept[AST_SIP_MAX_ACCEPT];
+ 
+    /*!
+     * \brief Called when a subscription is to be destroyed
+     *
+     * This is a subscriber and notifier callback.
+     *
+     * The handler is not expected to send any sort of requests or responses
+     * during this callback. The handler MUST, however, begin the destruction
+	 * process for the subscription during this callback.
+     */
+   void (*subscription_shutdown)(struct ast_sip_subscription *subscription);
+ 
+    /*!
+     * \brief Called when a SUBSCRIBE arrives in order to create a new subscription
+     *
+     * This is a notifier callback.
+     *
+     * If the notifier wishes to accept the subscription, then it can create
+     * a new ast_sip_subscription to do so. 
+     *
+     * If the notifier chooses to create a new subscription, then it must accept
+     * the incoming subscription using pjsip_evsub_accept() and it must also
+     * send an initial NOTIFY with the current subscription state.
+     *
+     * \param endpoint The endpoint from which we received the SUBSCRIBE
+     * \param rdata The SUBSCRIBE request
+     * \retval NULL The SUBSCRIBE has not been accepted
+     * \retval non-NULL The newly-created subscription
+     */
+    struct ast_sip_subscription *(*new_subscribe)(struct ast_sip_endpoint *endpoint,
+            pjsip_rx_data *rdata);
+ 
+    /*!
+     * \brief Called when an endpoint renews a subscription.
+     *
+     * This is a notifier callback.
+     *
+     * Because of the way that the PJSIP evsub framework works, it will automatically
+     * send a response to the SUBSCRIBE. However, the subscription handler must send
+	 * a NOTIFY with the current subscription state when this callback is called.
+	 *
+	 * The response_data that is passed into this callback is used to craft what should
+	 * be in the response to the incoming SUBSCRIBE. It is initialized with a 200 status
+	 * code and all other parameters are empty.
+     *
+     * \param sub The subscription that is being renewed
+     * \param rdata The SUBSCRIBE request in question
+	 * \param[out] response_data Data pertaining to the SIP response that should be
+	 * sent to the SUBSCRIBE
+     */
+    void (*resubscribe)(struct ast_sip_subscription *sub,
+            pjsip_rx_data *rdata, struct ast_sip_subscription_response_data *response_data);
+ 
+    /*!
+     * \brief Called when a subscription times out.
+     *
+     * This is a notifier callback
+     *
+     * This indicates that the subscription has timed out. The subscription handler is
+     * expected to send a NOTIFY that terminates the subscription.
+     *
+     * \param sub The subscription that has timed out
+     */
+    void (*subscription_timeout)(struct ast_sip_subscription *sub);
+ 
+    /*!
+     * \brief Called when a subscription is terminated via a SUBSCRIBE or NOTIFY request
+     *
+     * This is a notifier and subscriber callback.
+     *
+     * The PJSIP subscription framework will automatically send the response to the
+     * request. If a notifier receives this callback, then the subscription handler
+	 * is expected to send a final NOTIFY to terminate the subscription.
+     *
+     * \param sub The subscription being terminated
+     * \param rdata The request that terminated the subscription
+     */
+    void (*subscription_terminated)(struct ast_sip_subscription *sub, pjsip_rx_data *rdata);
+ 
+    /*!
+     * \brief Called when a subscription handler's outbound NOTIFY receives a response
+     *
+     * This is a notifier callback.
+     *
+     * \param sub The subscription
+     * \param rdata The NOTIFY response
+     */
+    void (*notify_response)(struct ast_sip_subscription *sub, pjsip_rx_data *rdata);
+ 
+    /*!
+     * \brief Called when a subscription handler receives an inbound NOTIFY
+     *
+     * This is a subscriber callback.
+     *
+     * Because of the way that the PJSIP evsub framework works, it will automatically
+     * send a response to the NOTIFY. By default this will be a 200 OK response, but
+     * this callback can change details of the response by returning response data
+     * to use.
+	 *
+	 * The response_data that is passed into this callback is used to craft what should
+	 * be in the response to the incoming SUBSCRIBE. It is initialized with a 200 status
+	 * code and all other parameters are empty.
+     *
+     * \param sub The subscription
+     * \param rdata The NOTIFY request
+	 * \param[out] response_data Data pertaining to the SIP response that should be
+	 * sent to the SUBSCRIBE
+     */
+    void (*notify_request)(struct ast_sip_subscription *sub,
+            pjsip_rx_data *rdata, struct ast_sip_subscription_response_data *response_data);
+ 
+    /*!
+     * \brief Called when it is time for a subscriber to resubscribe
+     *
+     * This is a subscriber callback.
+     *
+     * The subscriber can reresh the subscription using the pjsip_evsub_initiate()
+     * function.
+     *
+     * \param sub The subscription to refresh
+     * \retval 0 Success
+     * \retval non-zero Failure
+     */
+    int (*refresh_subscription)(struct ast_sip_subscription *sub);
+	AST_LIST_ENTRY(ast_sip_subscription_handler) next;
+};
+
+/*!
+ * \brief Create a new ast_sip_subscription structure
+ *
+ * In most cases the pubsub core will create a general purpose subscription
+ * within PJSIP. However, PJSIP provides enhanced support for the following
+ * event packages:
+ *
+ * presence
+ * message-summary
+ *
+ * If either of these events are handled by the subscription handler, then
+ * the special-purpose event subscriptions will be created within PJSIP,
+ * and it will be expected that your subscription handler make use of the
+ * special PJSIP APIs.
+ *
+ * \param handler The subsription handler for this subscription
+ * \param role Whether we are acting as subscriber or notifier for this subscription
+ * \param endpoint The endpoint involved in this subscription
+ * \param rdata If acting as a notifier, the SUBSCRIBE request that triggered subscription creation
+ */
+struct ast_sip_subscription *ast_sip_create_subscription(const struct ast_sip_subscription_handler *handler,
+        enum ast_sip_subscription_role role, struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata);
+ 
+ 
+/*!
+ * \brief Get the endpoint that is associated with this subscription
+ *
+ * This function will increase the reference count of the endpoint. Be sure to
+ * release the reference to it when you are finished with the endpoint.
+ *
+ * \retval NULL Could not get endpoint
+ * \retval non-NULL The endpoint
+ */
+struct ast_sip_endpoint *ast_sip_subscription_get_endpoint(struct ast_sip_subscription *sub);
+ 
+/*!
+ * \brief Get the serializer for the subscription
+ *
+ * Tasks that originate outside of a SIP servant thread should get the serializer
+ * and push the task to the serializer.
+ *
+ * \param sub The subscription
+ * \retval NULL Failure
+ * \retval non-NULL The subscription's serializer
+ */
+struct ast_taskprocessor *ast_sip_subscription_get_serializer(struct ast_sip_subscription *sub);
+ 
+/*!
+ * \brief Get the underlying PJSIP evsub structure
+ *
+ * This is useful when wishing to call PJSIP's API calls in order to
+ * create SUBSCRIBEs, NOTIFIES, etc. as well as get subscription state
+ *
+ * This function, as well as all methods called on the pjsip_evsub should
+ * be done in a SIP servant thread.
+ *
+ * \param sub The subscription
+ * \retval NULL Failure
+ * \retval non-NULL The underlying pjsip_evsub
+ */
+pjsip_evsub *ast_sip_subscription_get_evsub(struct ast_sip_subscription *sub);
+ 
+/*!
+ * \brief Send a request created via a PJSIP evsub method
+ *
+ * Callers of this function should take care to do so within a SIP servant
+ * thread.
+ *
+ * \param sub The subscription on which to send the request
+ * \param tdata The request to send
+ * \retval 0 Success
+ * \retval non-zero Failure
+ */
+int ast_sip_subscription_send_request(struct ast_sip_subscription *sub, pjsip_tx_data *tdata);
+ 
+/*!
+ * \brief Alternative for ast_datastore_alloc()
+ *
+ * There are two major differences between this and ast_datastore_alloc()
+ * 1) This allocates a refcounted object
+ * 2) This will fill in a uid if one is not provided
+ *
+ * DO NOT call ast_datastore_free() on a datastore allocated in this
+ * way since that function will attempt to free the datastore rather
+ * than play nicely with its refcount.
+ *
+ * \param info Callbacks for datastore
+ * \param uid Identifier for datastore
+ * \retval NULL Failed to allocate datastore
+ * \retval non-NULL Newly allocated datastore
+ */
+struct ast_datastore *ast_sip_subscription_alloc_datastore(const struct ast_datastore_info *info, const char *uid);
+ 
+/*!
+ * \brief Add a datastore to a SIP subscription
+ *
+ * Note that SIP uses reference counted datastores. The datastore passed into this function
+ * must have been allocated using ao2_alloc() or there will be serious problems.
+ *
+ * \param subscription The ssubscription to add the datastore to
+ * \param datastore The datastore to be added to the subscription
+ * \retval 0 Success
+ * \retval -1 Failure
+ */
+int ast_sip_subscription_add_datastore(struct ast_sip_subscription *subscription, struct ast_datastore *datastore);
+ 
+/*!
+ * \brief Retrieve a subscription datastore
+ *
+ * The datastore retrieved will have its reference count incremented. When the caller is done
+ * with the datastore, the reference counted needs to be decremented using ao2_ref().
+ *
+ * \param subscription The subscription from which to retrieve the datastore
+ * \param name The name of the datastore to retrieve
+ * \retval NULL Failed to find the specified datastore
+ * \retval non-NULL The specified datastore
+ */
+struct ast_datastore *ast_sip_subscription_get_datastore(struct ast_sip_subscription *subscription, const char *name);
+ 
+/*!
+ * \brief Remove a subscription datastore from the subscription
+ *
+ * This operation may cause the datastore's free() callback to be called if the reference
+ * count reaches zero.
+ *
+ * \param subscription The subscription to remove the datastore from
+ * \param name The name of the datastore to remove
+ */
+void ast_sip_subscription_remove_datastore(struct ast_sip_subscription *subscription, const char *name);
+ 
+/*!
+ * \brief Register a subscription handler
+ *
+ * \retval 0 Handler was registered successfully
+ * \retval non-zero Handler was not registered successfully
+ */
+int ast_sip_register_subscription_handler(struct ast_sip_subscription_handler *handler);
+ 
+/*!
+ * \brief Unregister a subscription handler
+ */
+void ast_sip_unregister_subscription_handler(struct ast_sip_subscription_handler *handler);
+
+#endif /* RES_SIP_PUBSUB_H */
diff --git a/include/asterisk/res_sip_session.h b/include/asterisk/res_sip_session.h
new file mode 100644
index 000000000..cbed52621
--- /dev/null
+++ b/include/asterisk/res_sip_session.h
@@ -0,0 +1,468 @@
+/*
+ * 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.
+ */
+
+#ifndef _RES_SIP_SESSION_H
+#define _RES_SIP_SESSION_H
+
+/* Needed for pj_timer_entry definition */
+#include "pjlib.h"
+#include "asterisk/linkedlists.h"
+/* Needed for AST_MAX_EXTENSION constant */
+#include "asterisk/channel.h"
+/* Needed for ast_sockaddr struct */
+#include "asterisk/netsock.h"
+
+/* Forward declarations */
+struct ast_sip_endpoint;
+struct ast_sip_transport;
+struct pjsip_inv_session;
+struct ast_channel;
+struct ast_datastore;
+struct ast_datastore_info;
+struct ao2_container;
+struct pjsip_tx_data;
+struct pjsip_rx_data;
+struct ast_party_id;
+struct pjmedia_sdp_media;
+struct pjmedia_sdp_session;
+struct ast_rtp_instance;
+
+struct ast_sip_session_sdp_handler;
+
+/*!
+ * \brief A structure containing SIP session media information
+ */
+struct ast_sip_session_media {
+	/*! \brief RTP instance itself */
+	struct ast_rtp_instance *rtp;
+	/*! \brief Direct media address */
+	struct ast_sockaddr direct_media_addr;
+	/*! \brief SDP handler that setup the RTP */
+	struct ast_sip_session_sdp_handler *handler;
+	/*! \brief Stream is on hold */
+	unsigned int held:1;
+	/*! \brief Stream type this session media handles */
+	char stream_type[1];
+};
+
+/*!
+ * \brief Opaque structure representing a request that could not be sent
+ * due to an outstanding INVITE transaction
+ */
+struct ast_sip_session_delayed_request;
+
+/*!
+ * \brief A structure describing a SIP session
+ *
+ * For the sake of brevity, a "SIP session" in Asterisk is referring to
+ * a dialog initiated by an INVITE. While "session" is typically interpreted
+ * to refer to the negotiated media within a SIP dialog, we have opted
+ * to use the term "SIP session" to refer to the INVITE dialog itself.
+ */
+struct ast_sip_session {
+	/* Dialplan extension where incoming call is destined */
+	char exten[AST_MAX_EXTENSION];
+	/* The endpoint with which Asterisk is communicating */
+	struct ast_sip_endpoint *endpoint;
+	/* The PJSIP details of the session, which includes the dialog */
+	struct pjsip_inv_session *inv_session;
+	/* The Asterisk channel associated with the session */
+	struct ast_channel *channel;
+	/* Registered session supplements */
+	AST_LIST_HEAD(, ast_sip_session_supplement) supplements;
+	/* Datastores added to the session by supplements to the session */
+	struct ao2_container *datastores;
+	/* Media streams */
+	struct ao2_container *media;
+	/* Serializer for tasks relating to this SIP session */
+	struct ast_taskprocessor *serializer;
+	/* Requests that could not be sent due to current inv_session state */
+	AST_LIST_HEAD_NOLOCK(, ast_sip_session_delayed_request) delayed_requests;
+	/* When we need to reschedule a reinvite, we use this structure to do it */
+	pj_timer_entry rescheduled_reinvite;
+	/* Format capabilities pertaining to direct media */
+	struct ast_format_cap *direct_media_cap;
+	/* Identity of endpoint this session deals with */
+	struct ast_party_id id;
+	/* Requested capabilities */
+	struct ast_format_cap *req_caps;
+};
+
+typedef int (*ast_sip_session_request_creation_cb)(struct ast_sip_session *session, pjsip_tx_data *tdata);
+typedef int (*ast_sip_session_response_cb)(struct ast_sip_session *session, pjsip_rx_data *rdata);
+
+enum ast_sip_session_supplement_priority {
+	/*! Top priority. Supplements with this priority are those that need to run before any others */
+	AST_SIP_SESSION_SUPPLEMENT_PRIORITY_FIRST = 0,
+	/*! Channel creation priority.
+	 * chan_gulp creates a channel at this priority. If your supplement depends on being run before
+	 * or after channel creation, then set your priority to be lower or higher than this value.
+	 */
+	AST_SIP_SESSION_SUPPLEMENT_PRIORITY_CHANNEL = 1000000,
+	/*! Lowest priority. Supplements with this priority should be run after all other supplements */
+	AST_SIP_SESSION_SUPPLEMENT_PRIORITY_LAST = INT_MAX,
+};
+
+/*!
+ * \brief A supplement to SIP message processing
+ *
+ * These can be registered by any module in order to add
+ * processing to incoming and outgoing SIP requests and responses
+ */
+struct ast_sip_session_supplement {
+    /*! Method on which to call the callbacks. If NULL, call on all methods */
+    const char *method;
+	/*! Priority for this supplement. Lower numbers are visited before higher numbers */
+	enum ast_sip_session_supplement_priority priority;
+    /*!
+	 * \brief Notification that the session has begun
+	 * This method will always be called from a SIP servant thread.
+	 */
+    void (*session_begin)(struct ast_sip_session *session);
+    /*! 
+	 * \brief Notification that the session has ended
+	 *
+	 * This method may or may not be called from a SIP servant thread. Do
+	 * not make assumptions about being able to call PJSIP methods from within
+	 * this method.
+	 */
+    void (*session_end)(struct ast_sip_session *session);
+	/*!
+	 * \brief Notification that the session is being destroyed
+	 */
+	void (*session_destroy)(struct ast_sip_session *session);
+    /*!
+     * \brief Called on incoming SIP request
+     * This method can indicate a failure in processing in its return. If there
+     * is a failure, it is required that this method sends a response to the request.
+	 * This method is always called from a SIP servant thread.
+	 *
+	 * \note
+	 * The following PJSIP methods will not work properly:
+	 * pjsip_rdata_get_dlg()
+	 * pjsip_rdata_get_tsx()
+	 * The reason is that the rdata passed into this function is a cloned rdata structure,
+	 * and its module data is not copied during the cloning operation.
+	 * If you need to get the dialog, you can get it via session->inv_session->dlg.
+     */
+    int (*incoming_request)(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
+    /*! 
+	 * \brief Called on an incoming SIP response
+	 * This method is always called from a SIP servant thread.
+	 *
+	 * \note
+	 * The following PJSIP methods will not work properly:
+	 * pjsip_rdata_get_dlg()
+	 * pjsip_rdata_get_tsx()
+	 * The reason is that the rdata passed into this function is a cloned rdata structure,
+	 * and its module data is not copied during the cloning operation.
+	 * If you need to get the dialog, you can get it via session->inv_session->dlg.
+	 */
+    void (*incoming_response)(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
+    /*!
+	 * \brief Called on an outgoing SIP request
+	 * This method is always called from a SIP servant thread.
+	 */
+    void (*outgoing_request)(struct ast_sip_session *session, struct pjsip_tx_data *tdata);
+    /*! 
+	 * \brief Called on an outgoing SIP response
+	 * This method is always called from a SIP servant thread.
+	 */
+    void (*outgoing_response)(struct ast_sip_session *session, struct pjsip_tx_data *tdata);
+	/*! Next item in the list */
+	AST_LIST_ENTRY(ast_sip_session_supplement) next;
+};
+
+/*!
+ * \brief A handler for SDPs in SIP sessions
+ *
+ * An SDP handler is registered by a module that is interested in being the
+ * responsible party for specific types of SDP streams.
+ */
+struct ast_sip_session_sdp_handler {
+	/*! An identifier for this handler */
+	const char *id;
+	/*!
+	 * \brief Set session details based on a stream in an incoming SDP offer or answer
+	 * \param session The session for which the media is being negotiated
+	 * \param session_media The media to be setup for this session
+	 * \param sdp The entire SDP. Useful for getting "global" information, such as connections or attributes
+	 * \param stream The stream on which to operate
+	 * \retval 0 The stream was not handled by this handler. If there are other registered handlers for this stream type, they will be called.
+	 * \retval <0 There was an error encountered. No further operation will take place and the current negotiation will be abandoned.
+	 * \retval >0 The stream was handled by this handler. No further handler of this stream type will be called.
+	 */
+	int (*negotiate_incoming_sdp_stream)(struct ast_sip_session *session, struct ast_sip_session_media *session_media, const struct pjmedia_sdp_session *sdp, const struct pjmedia_sdp_media *stream);
+	/*!
+	 * \brief Create an SDP media stream and add it to the outgoing SDP offer or answer
+	 * \param session The session for which media is being added
+	 * \param session_media The media to be setup for this session
+	 * \param stream The stream on which to operate
+	 * \retval 0 The stream was not handled by this handler. If there are other registered handlers for this stream type, they will be called.
+	 * \retval <0 There was an error encountered. No further operation will take place and the current negotiation will be abandoned.
+	 * \retval >0 The stream was handled by this handler. No further handler of this stream type will be called.
+	 */
+	int (*handle_incoming_sdp_stream)(struct ast_sip_session *session, struct ast_sip_session_media *session_media, const struct pjmedia_sdp_session *sdp, struct pjmedia_sdp_media *stream);
+	/*!
+	 * \brief Create an SDP media stream and add it to the outgoing SDP offer or answer
+	 * \param session The session for which media is being added
+	 * \param session_media The media to be setup for this session
+	 * \param sdp The entire SDP as currently built
+	 * \retval 0 This handler has no stream to add. If there are other registered handlers for this stream type, they will be called.
+	 * \retval <0 There was an error encountered. No further operation will take place and the current SDP negotiation will be abandoned.
+	 * \retval >0 The handler has a stream to be added to the SDP. No further handler of this stream type will be called.
+	 */
+	int (*create_outgoing_sdp_stream)(struct ast_sip_session *session, struct ast_sip_session_media *session_media, struct pjmedia_sdp_session *sdp);
+	/*!
+	 * \brief Update media stream with external address if applicable
+	 * \param tdata The outgoing message itself
+	 * \param stream The stream on which to operate
+	 * \param transport The transport the SDP is going out on
+	 */
+	void (*change_outgoing_sdp_stream_media_address)(struct pjsip_tx_data *tdata, struct pjmedia_sdp_media *stream, struct ast_sip_transport *transport);
+	/*!
+	 * \brief Apply a negotiated SDP media stream
+	 * \param session The session for which media is being applied
+	 * \param session_media The media to be setup for this session
+	 * \param local The entire local negotiated SDP
+	 * \param local_stream The local stream which to apply
+	 * \param remote The entire remote negotiated SDP
+	 * \param remote_stream The remote stream which to apply
+	 * \retval 0 The stream was not applied by this handler. If there are other registered handlers for this stream type, they will be called.
+	 * \retval <0 There was an error encountered. No further operation will take place and the current application will be abandoned.
+	 * \retval >0 The stream was handled by this handler. No further handler of this stream type will be called.
+	 */
+	int (*apply_negotiated_sdp_stream)(struct ast_sip_session *session, struct ast_sip_session_media *session_media, const struct pjmedia_sdp_session *local, const struct pjmedia_sdp_media *local_stream,
+		const struct pjmedia_sdp_session *remote, const struct pjmedia_sdp_media *remote_stream);
+	/*!
+	 * \brief Destroy a session_media created by this handler
+	 * \param session The session for which media is being destroyed
+	 * \param session_media The media to destroy
+	 */
+	void (*stream_destroy)(struct ast_sip_session_media *session_media);
+	/*! Next item in the list. */
+	AST_LIST_ENTRY(ast_sip_session_sdp_handler) next;
+};
+
+/*!
+ * \brief Allocate a new SIP session
+ *
+ * This will take care of allocating the datastores container on the session as well
+ * as placing all registered supplements onto the session.
+ *
+ * The endpoint that is passed in will have its reference count increased by one since
+ * the session will be keeping a reference to the endpoint. The session will relinquish
+ * this reference when the session is destroyed.
+ *
+ * \param endpoint The endpoint that this session communicates with
+ * \param inv_session The PJSIP INVITE session data
+ */
+struct ast_sip_session *ast_sip_session_alloc(struct ast_sip_endpoint *endpoint, pjsip_inv_session *inv);
+
+/*!
+ * \brief Create a new outgoing SIP session
+ *
+ * The endpoint that is passed in will have its reference count increased by one since
+ * the session will be keeping a reference to the endpoint. The session will relinquish
+ * this reference when the session is destroyed.
+ *
+ * \param endpoint The endpoint that this session uses for settings
+ * \param location Optional name of the location to call, be it named location or explicit URI
+ * \param request_user Optional request user to place in the request URI if permitted
+ * \param req_caps The requested capabilities
+ */
+struct ast_sip_session *ast_sip_session_create_outgoing(struct ast_sip_endpoint *endpoint, const char *location, const char *request_user, struct ast_format_cap *req_caps);
+
+/*!
+ * \brief Register an SDP handler
+ *
+ * An SDP handler is responsible for parsing incoming SDP streams and ensuring that
+ * Asterisk can cope with the contents. Similarly, the SDP handler will be
+ * responsible for constructing outgoing SDP streams.
+ *
+ * Multiple handlers for the same stream type may be registered. They will be
+ * visited in the order they were registered. Handlers will be visited for each
+ * stream type until one claims to have handled the stream.
+ *
+ * \param handler The SDP handler to register
+ * \param stream_type The type of media stream for which to call the handler
+ * \retval 0 Success
+ * \retval -1 Failure
+ */
+int ast_sip_session_register_sdp_handler(struct ast_sip_session_sdp_handler *handler, const char *stream_type);
+
+/*!
+ * \brief Unregister an SDP handler
+ *
+ * \param handler The SDP handler to unregister
+ * \param stream_type Stream type for which the SDP handler was registered
+ */
+void ast_sip_session_unregister_sdp_handler(struct ast_sip_session_sdp_handler *handler, const char *stream_type);
+
+/*!
+ * \brief Register a supplement to SIP session processing
+ *
+ * This allows for someone to insert themselves in the processing of SIP
+ * requests and responses. This, for example could allow for a module to
+ * set channel data based on headers in an incoming message. Similarly,
+ * a module could reject an incoming request if desired.
+ *
+ * \param supplement The supplement to register
+ * \retval 0 Success
+ * \retval -1 Failure
+ */
+int ast_sip_session_register_supplement(struct ast_sip_session_supplement *supplement);
+
+/*!
+ * \brief Unregister a an supplement to SIP session processing
+ *
+ * \param supplement The supplement to unregister
+ */
+void ast_sip_session_unregister_supplement(struct ast_sip_session_supplement *supplement);
+
+/*!
+ * \brief Alternative for ast_datastore_alloc()
+ *
+ * There are two major differences between this and ast_datastore_alloc()
+ * 1) This allocates a refcounted object
+ * 2) This will fill in a uid if one is not provided
+ *
+ * DO NOT call ast_datastore_free() on a datastore allocated in this
+ * way since that function will attempt to free the datastore rather
+ * than play nicely with its refcount.
+ *
+ * \param info Callbacks for datastore
+ * \param uid Identifier for datastore
+ * \retval NULL Failed to allocate datastore
+ * \retval non-NULL Newly allocated datastore
+ */
+struct ast_datastore *ast_sip_session_alloc_datastore(const struct ast_datastore_info *info, const char *uid);
+
+/*!
+ * \brief Add a datastore to a SIP session
+ *
+ * Note that SIP uses reference counted datastores. The datastore passed into this function
+ * must have been allocated using ao2_alloc() or there will be serious problems.
+ *
+ * \param session The session to add the datastore to
+ * \param datastore The datastore to be added to the session
+ * \retval 0 Success
+ * \retval -1 Failure
+ */
+int ast_sip_session_add_datastore(struct ast_sip_session *session, struct ast_datastore *datastore);
+
+/*!
+ * \brief Retrieve a session datastore
+ *
+ * The datastore retrieved will have its reference count incremented. When the caller is done
+ * with the datastore, the reference counted needs to be decremented using ao2_ref().
+ *
+ * \param session The session from which to retrieve the datastore
+ * \param name The name of the datastore to retrieve
+ * \retval NULL Failed to find the specified datastore
+ * \retval non-NULL The specified datastore
+ */
+struct ast_datastore *ast_sip_session_get_datastore(struct ast_sip_session *session, const char *name);
+
+/*!
+ * \brief Remove a session datastore from the session
+ *
+ * This operation may cause the datastore's free() callback to be called if the reference
+ * count reaches zero.
+ *
+ * \param session The session to remove the datastore from
+ * \param name The name of the datastore to remove
+ */
+void ast_sip_session_remove_datastore(struct ast_sip_session *session, const char *name);
+
+/*!
+ * \brief Retrieve identifying information from an incoming request
+ *
+ * This will retrieve identifying information and place it in the
+ * id parameter. The caller of the function can then apply this to
+ * caller ID, connected line, or whatever else may be proper.
+ *
+ * \param rdata The incoming request or response
+ * \param[out] id The collected identity information
+ * \retval 0 Successfully found identifying information
+ * \retval -1 Identifying information could not be found
+ */
+int ast_sip_session_get_identity(struct pjsip_rx_data *rdata, struct ast_party_id *id);
+
+/*!
+ * \brief Send a reinvite or UPDATE on a session
+ *
+ * This method will inspect the session in order to construct an appropriate
+ * session refresh request. As with any outgoing request in res_sip_session,
+ * this will call into registered supplements in case they wish to add anything.
+ *
+ * Note: The on_request_creation callback may or may not be called in the same
+ * thread where this function is called. Request creation may need to be delayed
+ * due to the current INVITE transaction state.
+ * 
+ * \param session The session on which the reinvite will be sent
+ * \param on_request_creation Callback called when request is created
+ * \param on_response Callback called when response for request is received
+ * \param method The method that should be used when constructing the session refresh
+ * \param generate_new_sdp Boolean to indicate if a new SDP should be created
+ * \retval 0 Successfully sent refresh
+ * \retval -1 Failure to send refresh
+ */
+int ast_sip_session_refresh(struct ast_sip_session *session,
+		ast_sip_session_request_creation_cb on_request_creation,
+		ast_sip_session_response_cb on_response,
+		enum ast_sip_session_refresh_method method,
+		int generate_new_sdp);
+
+/*!
+ * \brief Send a SIP response
+ *
+ * This will send the SIP response specified in tdata and
+ * call into any registered supplements' outgoing_response callback.
+ *
+ * \param session The session on which to send the response.
+ * \param tdata The response to send
+ */
+void ast_sip_session_send_response(struct ast_sip_session *session, pjsip_tx_data *tdata);
+
+/*!
+ * \brief Send a SIP request
+ *
+ * This will send the SIP request specified in tdata and
+ * call into any registered supplements' outgoing_request callback.
+ *
+ * \param session The session to which to send the request
+ * \param tdata The request to send
+ */
+void ast_sip_session_send_request(struct ast_sip_session *session, pjsip_tx_data *tdata);
+
+/*!
+ * \brief Send a SIP request and get called back when a response is received
+ *
+ * This will send the request out exactly the same as ast_sip_send_request() does.
+ * The difference is that when a response arrives, the specified callback will be
+ * called into
+ *
+ * \param session The session on which to send the request
+ * \param tdata The request to send
+ * \param on_response Callback to be called when a response is received
+ */
+void ast_sip_session_send_request_with_cb(struct ast_sip_session *session, pjsip_tx_data *tdata,
+		ast_sip_session_response_cb on_response);
+
+#endif /* _RES_SIP_SESSION_H */
diff --git a/include/asterisk/sorcery.h b/include/asterisk/sorcery.h
index e390b43cf..434f5595a 100644
--- a/include/asterisk/sorcery.h
+++ b/include/asterisk/sorcery.h
@@ -157,10 +157,15 @@ typedef struct ast_variable *(*sorcery_transform_handler)(struct ast_variable *s
 /*!
  * \brief A callback function for when an object set is successfully applied to an object
  *
+ * \note On a failure return, the state of the object is left undefined. It is a bad
+ * idea to try to use this object.
+ *
  * \param sorcery Sorcery structure in use
  * \param obj The object itself
+ * \retval 0 Success
+ * \retval non-zero Failure
  */
-typedef void (*sorcery_apply_handler)(const struct ast_sorcery *sorcery, void *obj);
+typedef int (*sorcery_apply_handler)(const struct ast_sorcery *sorcery, void *obj);
 
 /*!
  * \brief A callback function for copying the contents of one object to another
diff --git a/include/asterisk/threadpool.h b/include/asterisk/threadpool.h
index 89076265e..e1e7727f5 100644
--- a/include/asterisk/threadpool.h
+++ b/include/asterisk/threadpool.h
@@ -108,6 +108,20 @@ struct ast_threadpool_options {
 	 * maximum size.
 	 */
 	int max_size;
+	/*!
+	 * \brief Function to call when a thread starts
+	 *
+	 * This is useful if there is something common that all threads
+	 * in a threadpool need to do when they start.
+	 */
+	void (*thread_start)(void);
+	/*!
+	 * \brief Function to call when a thread ends
+	 *
+	 * This is useful if there is common cleanup to execute when
+	 * a thread completes
+	 */
+	void (*thread_end)(void);
 };
 
 /*!