asterisk/channels/chan_motif.c

/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2012, Digium, Inc.
 *
 * Joshua Colp <jcolp@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.
 */

/*! \file
 *
 * \author Joshua Colp <jcolp@digium.com>
 *
 * \brief Motif Jingle Channel Driver
 *
 * \extref Iksemel http://iksemel.jabberstudio.org/
 *
 * \ingroup channel_drivers
 */

/*** MODULEINFO
	<depend>iksemel</depend>
	<depend>res_xmpp</depend>
	<use type="external">openssl</use>
	<support_level>core</support_level>
 ***/

#include "asterisk.h"

ASTERISK_FILE_VERSION(__FILE__, "$Revision$")

#include <sys/socket.h>
#include <fcntl.h>
#include <netdb.h>
#include <netinet/in.h>
#include <arpa/inet.h>
#include <sys/signal.h>
#include <iksemel.h>
#include <pthread.h>

#include "asterisk/lock.h"
#include "asterisk/channel.h"
#include "asterisk/config_options.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/acl.h"
#include "asterisk/callerid.h"
#include "asterisk/file.h"
#include "asterisk/cli.h"
#include "asterisk/app.h"
#include "asterisk/musiconhold.h"
#include "asterisk/manager.h"
#include "asterisk/stringfields.h"
#include "asterisk/utils.h"
#include "asterisk/causes.h"
#include "asterisk/astobj.h"
#include "asterisk/abstract_jb.h"
#include "asterisk/xmpp.h"

/*! \brief Default maximum number of ICE candidates we will offer */
#define DEFAULT_MAX_ICE_CANDIDATES "10"

/*! \brief Default maximum number of payloads we will offer */
#define DEFAULT_MAX_PAYLOADS "30"

/*! \brief Number of buckets for endpoints */
#define ENDPOINT_BUCKETS 37

/*! \brief Number of buckets for sessions, on a per-endpoint basis */
#define SESSION_BUCKETS 37

/*! \brief Namespace for Jingle itself */
#define JINGLE_NS "urn:xmpp:jingle:1"

/*! \brief Namespace for Jingle RTP sessions */
#define JINGLE_RTP_NS "urn:xmpp:jingle:apps:rtp:1"

/*! \brief Namespace for Jingle RTP info */
#define JINGLE_RTP_INFO_NS "urn:xmpp:jingle:apps:rtp:info:1"

/*! \brief Namespace for Jingle ICE-UDP */
#define JINGLE_ICE_UDP_NS "urn:xmpp:jingle:transports:ice-udp:1"

/*! \brief Namespace for Google Talk ICE-UDP */
#define GOOGLE_TRANSPORT_NS "http://www.google.com/transport/p2p"

/*! \brief Namespace for Google Talk Raw UDP */
#define GOOGLE_TRANSPORT_RAW_NS "http://www.google.com/transport/raw-udp"

/*! \brief Namespace for Google Session */
#define GOOGLE_SESSION_NS "http://www.google.com/session"

/*! \brief Namespace for Google Phone description */
#define GOOGLE_PHONE_NS "http://www.google.com/session/phone"

/*! \brief Namespace for Google Video description */
#define GOOGLE_VIDEO_NS "http://www.google.com/session/video"

/*! \brief Namespace for XMPP stanzas */
#define XMPP_STANZAS_NS "urn:ietf:params:xml:ns:xmpp-stanzas"

/*! \brief The various transport methods supported, from highest priority to lowest priority when doing fallback */
enum jingle_transport {
	JINGLE_TRANSPORT_ICE_UDP = 3,   /*!< XEP-0176 */
	JINGLE_TRANSPORT_GOOGLE_V2 = 2, /*!< https://developers.google.com/talk/call_signaling */
	JINGLE_TRANSPORT_GOOGLE_V1 = 1, /*!< Undocumented initial Google specification */
	JINGLE_TRANSPORT_NONE = 0,      /*!< No transport specified */
};

/*! \brief Endpoint state information */
struct jingle_endpoint_state {
	struct ao2_container *sessions; /*!< Active sessions to or from the endpoint */
};

/*! \brief Endpoint which contains configuration information and active sessions */
struct jingle_endpoint {
	AST_DECLARE_STRING_FIELDS(
		AST_STRING_FIELD(name);              /*!< Name of the endpoint */
		AST_STRING_FIELD(context);           /*!< Context to place incoming calls into */
		AST_STRING_FIELD(accountcode);       /*!< Account code */
		AST_STRING_FIELD(language);          /*!< Default language for prompts */
		AST_STRING_FIELD(musicclass);        /*!< Configured music on hold class */
		AST_STRING_FIELD(parkinglot);        /*!< Configured parking lot */
		);
	struct ast_xmpp_client *connection;     /*!< Connection to use for traffic */
	iksrule *rule;                          /*!< Active matching rule */
	unsigned int maxicecandidates;          /*!< Maximum number of ICE candidates we will offer */
	unsigned int maxpayloads;               /*!< Maximum number of payloads we will offer */
	struct ast_codec_pref prefs;            /*!< Codec preferences */
	struct ast_format_cap *cap;             /*!< Formats to use */
	ast_group_t callgroup;                  /*!< Call group */
	ast_group_t pickupgroup;                /*!< Pickup group */
	enum jingle_transport transport;        /*!< Default transport to use on outgoing sessions */
	struct jingle_endpoint_state *state;    /*!< Endpoint state information */
};

/*! \brief Session which contains information about an active session */
struct jingle_session {
	AST_DECLARE_STRING_FIELDS(
		AST_STRING_FIELD(sid);        /*!< Session identifier */
		AST_STRING_FIELD(audio_name); /*!< Name of the audio content */
		AST_STRING_FIELD(video_name); /*!< Name of the video content */
		);
	struct jingle_endpoint_state *state;  /*!< Endpoint we are associated with */
	struct ast_xmpp_client *connection;   /*!< Connection to use for traffic */
	enum jingle_transport transport;      /*!< Transport type to use for this session */
	unsigned int maxicecandidates;        /*!< Maximum number of ICE candidates we will offer */
	unsigned int maxpayloads;             /*!< Maximum number of payloads we will offer */
	char remote_original[XMPP_MAX_JIDLEN];/*!< Identifier of the original remote party (remote may have changed due to redirect) */
	char remote[XMPP_MAX_JIDLEN];         /*!< Identifier of the remote party */
	iksrule *rule;                        /*!< Session matching rule */
	struct ast_codec_pref prefs;          /*!< Codec preferences */
	struct ast_channel *owner;            /*!< Master Channel */
	struct ast_rtp_instance *rtp;         /*!< RTP audio session */
	struct ast_rtp_instance *vrtp;        /*!< RTP video session */
	struct ast_format_cap *cap;           /*!< Local codec capabilities */
	struct ast_format_cap *jointcap;      /*!< Joint codec capabilities */
	struct ast_format_cap *peercap;       /*!< Peer codec capabilities */
	unsigned int outgoing:1;              /*!< Whether this is an outgoing leg or not */
	unsigned int gone:1;                  /*!< In the eyes of Jingle this session is already gone */
	struct ast_callid *callid;            /*!< Bound session call-id */
};

static const char desc[] = "Motif Jingle Channel";
static const char channel_type[] = "Motif";

struct jingle_config {
	struct ao2_container *endpoints; /*!< Configured endpoints */
};

static AO2_GLOBAL_OBJ_STATIC(globals);

static struct ast_sched_context *sched; /*!< Scheduling context for RTCP */

/* \brief Asterisk core interaction functions */
static struct ast_channel *jingle_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, const char *data, int *cause);
static int jingle_sendtext(struct ast_channel *ast, const char *text);
static int jingle_digit_begin(struct ast_channel *ast, char digit);
static int jingle_digit_end(struct ast_channel *ast, char digit, unsigned int duration);
static int jingle_call(struct ast_channel *ast, const char *dest, int timeout);
static int jingle_hangup(struct ast_channel *ast);
static int jingle_answer(struct ast_channel *ast);
static struct ast_frame *jingle_read(struct ast_channel *ast);
static int jingle_write(struct ast_channel *ast, struct ast_frame *f);
static int jingle_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen);
static int jingle_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
static struct jingle_session *jingle_alloc(struct jingle_endpoint *endpoint, const char *from, const char *sid);

/*! \brief Action handlers */
static void jingle_action_session_initiate(struct jingle_endpoint *endpoint, struct jingle_session *session, ikspak *pak);
static void jingle_action_transport_info(struct jingle_endpoint *endpoint, struct jingle_session *session, ikspak *pak);
static void jingle_action_session_accept(struct jingle_endpoint *endpoint, struct jingle_session *session, ikspak *pak);
static void jingle_action_session_info(struct jingle_endpoint *endpoint, struct jingle_session *session, ikspak *pak);
static void jingle_action_session_terminate(struct jingle_endpoint *endpoint, struct jingle_session *session, ikspak *pak);

/*! \brief PBX interface structure for channel registration */
static struct ast_channel_tech jingle_tech = {
	.type = "Motif",
	.description = "Motif Jingle Channel Driver",
	.requester = jingle_request,
	.send_text = jingle_sendtext,
	.send_digit_begin = jingle_digit_begin,
	.send_digit_end = jingle_digit_end,
	.bridge = ast_rtp_instance_bridge,
	.call = jingle_call,
	.hangup = jingle_hangup,
	.answer = jingle_answer,
	.read = jingle_read,
	.write = jingle_write,
	.write_video = jingle_write,
	.exception = jingle_read,
	.indicate = jingle_indicate,
	.fixup = jingle_fixup,
	.properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER
};

/*! \brief Defined handlers for different Jingle actions */
static const struct jingle_action_handler {
	const char *action;
	void (*handler)(struct jingle_endpoint *endpoint, struct jingle_session *session, ikspak *pak);
} jingle_action_handlers[] = {
	/* Jingle actions */
	{ "session-initiate", jingle_action_session_initiate, },
	{ "transport-info", jingle_action_transport_info, },
	{ "session-accept", jingle_action_session_accept, },
	{ "session-info", jingle_action_session_info, },
	{ "session-terminate", jingle_action_session_terminate, },
	/* Google-V1 actions */
	{ "initiate", jingle_action_session_initiate, },
	{ "candidates", jingle_action_transport_info, },
	{ "accept", jingle_action_session_accept, },
	{ "terminate", jingle_action_session_terminate, },
	{ "reject", jingle_action_session_terminate, },
};

/*! \brief Reason text <-> cause code mapping */
static const struct jingle_reason_mapping {
	const char *reason;
	int cause;
} jingle_reason_mappings[] = {
	{ "busy", AST_CAUSE_BUSY, },
	{ "cancel", AST_CAUSE_CALL_REJECTED, },
	{ "connectivity-error", AST_CAUSE_INTERWORKING, },
	{ "decline", AST_CAUSE_CALL_REJECTED, },
	{ "expired", AST_CAUSE_NO_USER_RESPONSE, },
	{ "failed-transport", AST_CAUSE_PROTOCOL_ERROR, },
	{ "failed-application", AST_CAUSE_SWITCH_CONGESTION, },
	{ "general-error", AST_CAUSE_CONGESTION, },
	{ "gone", AST_CAUSE_NORMAL_CLEARING, },
	{ "incompatible-parameters", AST_CAUSE_BEARERCAPABILITY_NOTAVAIL, },
	{ "media-error", AST_CAUSE_BEARERCAPABILITY_NOTAVAIL, },
	{ "security-error", AST_CAUSE_PROTOCOL_ERROR, },
	{ "success", AST_CAUSE_NORMAL_CLEARING, },
	{ "timeout", AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE, },
	{ "unsupported-applications", AST_CAUSE_BEARERCAPABILITY_NOTAVAIL, },
	{ "unsupported-transports", AST_CAUSE_FACILITY_NOT_IMPLEMENTED, },
};

/*! \brief Hashing function for Jingle sessions */
static int jingle_session_hash(const void *obj, const int flags)
{
	const struct jingle_session *session = obj;
	const char *sid = obj;

	return ast_str_hash(flags & OBJ_KEY ? sid : session->sid);
}

/*! \brief Comparator function for Jingle sessions */
static int jingle_session_cmp(void *obj, void *arg, int flags)
{
	struct jingle_session *session1 = obj, *session2 = arg;
	const char *sid = arg;

	return !strcmp(session1->sid, flags & OBJ_KEY ? sid : session2->sid) ? CMP_MATCH | CMP_STOP : 0;
}

/*! \brief Destructor for Jingle endpoint state */
static void jingle_endpoint_state_destructor(void *obj)
{
	struct jingle_endpoint_state *state = obj;

	ao2_ref(state->sessions, -1);
}

/*! \brief Destructor for Jingle endpoints */
static void jingle_endpoint_destructor(void *obj)
{
	struct jingle_endpoint *endpoint = obj;

	if (endpoint->rule) {
		iks_filter_remove_rule(endpoint->connection->filter, endpoint->rule);
	}

	if (endpoint->connection) {
		ast_xmpp_client_unref(endpoint->connection);
	}

	ast_format_cap_destroy(endpoint->cap);

	ao2_ref(endpoint->state, -1);

	ast_string_field_free_memory(endpoint);
}

/*! \brief Find function for Jingle endpoints */
static void *jingle_endpoint_find(struct ao2_container *tmp_container, const char *category)
{
	return ao2_find(tmp_container, category, OBJ_KEY);
}

/*! \brief Allocator function for Jingle endpoint state */
static struct jingle_endpoint_state *jingle_endpoint_state_create(void)
{
	struct jingle_endpoint_state *state;

	if (!(state = ao2_alloc(sizeof(*state), jingle_endpoint_state_destructor))) {
		return NULL;
	}

	if (!(state->sessions = ao2_container_alloc(SESSION_BUCKETS, jingle_session_hash, jingle_session_cmp))) {
		ao2_ref(state, -1);
		return NULL;
	}

	return state;
}

/*! \brief State find/create function */
static struct jingle_endpoint_state *jingle_endpoint_state_find_or_create(const char *category)
{
	RAII_VAR(struct jingle_config *, cfg, ao2_global_obj_ref(globals), ao2_cleanup);
	RAII_VAR(struct jingle_endpoint *, endpoint, NULL, ao2_cleanup);

	if (!cfg || !cfg->endpoints || !(endpoint = jingle_endpoint_find(cfg->endpoints, category))) {
		return jingle_endpoint_state_create();
	}

	ao2_ref(endpoint->state, +1);
	return endpoint->state;
}

/*! \brief Allocator function for Jingle endpoints */
static void *jingle_endpoint_alloc(const char *cat)
{
	struct jingle_endpoint *endpoint;

	if (!(endpoint = ao2_alloc(sizeof(*endpoint), jingle_endpoint_destructor))) {
		return NULL;
	}

	if (ast_string_field_init(endpoint, 512)) {
		ao2_ref(endpoint, -1);
		return NULL;
	}

	if (!(endpoint->state = jingle_endpoint_state_find_or_create(cat))) {
		ao2_ref(endpoint, -1);
		return NULL;
	}

	ast_string_field_set(endpoint, name, cat);

	endpoint->cap = ast_format_cap_alloc_nolock();
	endpoint->transport = JINGLE_TRANSPORT_ICE_UDP;

	return endpoint;
}

/*! \brief Hashing function for Jingle endpoints */
static int jingle_endpoint_hash(const void *obj, const int flags)
{
	const struct jingle_endpoint *endpoint = obj;
	const char *name = obj;

	return ast_str_hash(flags & OBJ_KEY ? name : endpoint->name);
}

/*! \brief Comparator function for Jingle endpoints */
static int jingle_endpoint_cmp(void *obj, void *arg, int flags)
{
	struct jingle_endpoint *endpoint1 = obj, *endpoint2 = arg;
	const char *name = arg;

	return !strcmp(endpoint1->name, flags & OBJ_KEY ? name : endpoint2->name) ? CMP_MATCH | CMP_STOP : 0;
}

static struct aco_type endpoint_option = {
	.type = ACO_ITEM,
	.category_match = ACO_BLACKLIST,
	.category = "^general$",
	.item_alloc = jingle_endpoint_alloc,
	.item_find = jingle_endpoint_find,
	.item_offset = offsetof(struct jingle_config, endpoints),
};

struct aco_type *endpoint_options[] = ACO_TYPES(&endpoint_option);

struct aco_file jingle_conf = {
	.filename = "motif.conf",
	.types = ACO_TYPES(&endpoint_option),
};

/*! \brief Destructor for Jingle sessions */
static void jingle_session_destructor(void *obj)
{
	struct jingle_session *session = obj;

	if (session->rule) {
		iks_filter_remove_rule(session->connection->filter, session->rule);
	}

	if (session->connection) {
		ast_xmpp_client_unref(session->connection);
	}

	if (session->rtp) {
		ast_rtp_instance_stop(session->rtp);
		ast_rtp_instance_destroy(session->rtp);
	}

	if (session->vrtp) {
		ast_rtp_instance_stop(session->vrtp);
		ast_rtp_instance_destroy(session->vrtp);
	}

	ast_format_cap_destroy(session->cap);
	ast_format_cap_destroy(session->jointcap);
	ast_format_cap_destroy(session->peercap);

	if (session->callid) {
		ast_callid_unref(session->callid);
	}

	ast_string_field_free_memory(session);
}

/*! \brief Destructor called when module configuration goes away */
static void jingle_config_destructor(void *obj)
{
	struct jingle_config *cfg = obj;
	ao2_cleanup(cfg->endpoints);
}

/*! \brief Allocator called when module configuration should appear */
static void *jingle_config_alloc(void)
{
	struct jingle_config *cfg;

	if (!(cfg = ao2_alloc(sizeof(*cfg), jingle_config_destructor))) {
		return NULL;
	}

	if (!(cfg->endpoints = ao2_container_alloc(ENDPOINT_BUCKETS, jingle_endpoint_hash, jingle_endpoint_cmp))) {
		ao2_ref(cfg, -1);
		return NULL;
	}

	return cfg;
}

CONFIG_INFO_STANDARD(cfg_info, globals, jingle_config_alloc,
		     .files = ACO_FILES(&jingle_conf),
	);

/*! \brief Function called by RTP engine to get local RTP peer */
static enum ast_rtp_glue_result jingle_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
	struct jingle_session *session = ast_channel_tech_pvt(chan);
	enum ast_rtp_glue_result res = AST_RTP_GLUE_RESULT_LOCAL;

	if (!session->rtp) {
		return AST_RTP_GLUE_RESULT_FORBID;
	}

	ao2_ref(session->rtp, +1);
	*instance = session->rtp;

	return res;
}

/*! \brief Function called by RTP engine to get peer capabilities */
static void jingle_get_codec(struct ast_channel *chan, struct ast_format_cap *result)
{
}

/*! \brief Function called by RTP engine to change where the remote party should send media */
static int jingle_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, struct ast_rtp_instance *tpeer, const struct ast_format_cap *cap, int nat_active)
{
	return -1;
}

/*! \brief Local glue for interacting with the RTP engine core */
static struct ast_rtp_glue jingle_rtp_glue = {
	.type = "Motif",
	.get_rtp_info = jingle_get_rtp_peer,
	.get_codec = jingle_get_codec,
	.update_peer = jingle_set_rtp_peer,
};

/*! \brief Internal helper function which enables video support on a sesson if possible */
static void jingle_enable_video(struct jingle_session *session)
{
	struct ast_sockaddr tmp;
	struct ast_rtp_engine_ice *ice;

	/* If video is already present don't do anything */
	if (session->vrtp) {
		return;
	}

	/* If there are no configured video codecs do not turn video support on, it just won't work */
	if (!ast_format_cap_has_type(session->cap, AST_FORMAT_TYPE_VIDEO)) {
		return;
	}

	ast_sockaddr_parse(&tmp, "0.0.0.0", 0);

	if (!(session->vrtp = ast_rtp_instance_new("asterisk", sched, &tmp, NULL))) {
		return;
	}

	ast_rtp_instance_set_prop(session->vrtp, AST_RTP_PROPERTY_RTCP, 1);

	ast_channel_set_fd(session->owner, 2, ast_rtp_instance_fd(session->vrtp, 0));
	ast_channel_set_fd(session->owner, 3, ast_rtp_instance_fd(session->vrtp, 1));
	ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(session->vrtp), session->vrtp, &session->prefs);

	if (session->transport == JINGLE_TRANSPORT_GOOGLE_V2 && (ice = ast_rtp_instance_get_ice(session->vrtp))) {
		ice->stop(session->vrtp);
	}
}

/*! \brief Internal helper function used to allocate Jingle session on an endpoint */
static struct jingle_session *jingle_alloc(struct jingle_endpoint *endpoint, const char *from, const char *sid)
{
	struct jingle_session *session;
	struct ast_callid *callid;
	struct ast_sockaddr tmp;

	if (!(session = ao2_alloc(sizeof(*session), jingle_session_destructor))) {
		return NULL;
	}

	callid = ast_read_threadstorage_callid();
	session->callid = (callid ? callid : ast_create_callid());

	if (ast_string_field_init(session, 512)) {
		ao2_ref(session, -1);
		return NULL;
	}

	if (!ast_strlen_zero(from)) {
		ast_copy_string(session->remote_original, from, sizeof(session->remote_original));
		ast_copy_string(session->remote, from, sizeof(session->remote));
	}

	if (ast_strlen_zero(sid)) {
		ast_string_field_build(session, sid, "%08lx%08lx", (unsigned long)ast_random(), (unsigned long)ast_random());
		session->outgoing = 1;
		ast_string_field_set(session, audio_name, "audio");
		ast_string_field_set(session, video_name, "video");
	} else {
		ast_string_field_set(session, sid, sid);
	}

	ao2_ref(endpoint->state, +1);
	session->state = endpoint->state;
	ao2_ref(endpoint->connection, +1);
	session->connection = endpoint->connection;
	session->transport = endpoint->transport;

	if (!(session->cap = ast_format_cap_alloc_nolock()) ||
	    !(session->jointcap = ast_format_cap_alloc_nolock()) ||
	    !(session->peercap = ast_format_cap_alloc_nolock()) ||
	    !session->callid) {
		ao2_ref(session, -1);
		return NULL;
	}

	ast_format_cap_copy(session->cap, endpoint->cap);

	/* While we rely on res_xmpp for communication we still need a temporary ast_sockaddr to tell the RTP engine
	 * that we want IPv4 */
	ast_sockaddr_parse(&tmp, "0.0.0.0", 0);

	/* Sessions always carry audio, but video is optional so don't enable it here */
	if (!(session->rtp = ast_rtp_instance_new("asterisk", sched, &tmp, NULL))) {
		ao2_ref(session, -1);
		return NULL;
	}
	ast_rtp_instance_set_prop(session->rtp, AST_RTP_PROPERTY_RTCP, 1);
	ast_rtp_instance_set_prop(session->rtp, AST_RTP_PROPERTY_DTMF, 1);

	memcpy(&session->prefs, &endpoint->prefs, sizeof(session->prefs));

	session->maxicecandidates = endpoint->maxicecandidates;
	session->maxpayloads = endpoint->maxpayloads;

	return session;
}

/*! \brief Function called to create a new Jingle Asterisk channel */
static struct ast_channel *jingle_new(struct jingle_endpoint *endpoint, struct jingle_session *session, int state, const char *title, const char *linkedid, const char *cid_name)
{
	struct ast_channel *chan;
	const char *str = S_OR(title, session->remote);
	struct ast_format tmpfmt;

	if (ast_format_cap_is_empty(session->cap)) {
		return NULL;
	}

	if (!(chan = ast_channel_alloc(1, state, S_OR(title, ""), S_OR(cid_name, ""), "", "", "", linkedid, 0, "Motif/%s-%04lx", str, (unsigned long)(ast_random() & 0xffff)))) {
		return NULL;
	}

	ast_channel_tech_set(chan, &jingle_tech);
	ast_channel_tech_pvt_set(chan, session);
	session->owner = chan;

	ast_channel_callid_set(chan, session->callid);

	ast_format_cap_copy(ast_channel_nativeformats(chan), session->cap);
	ast_codec_choose(&session->prefs, session->cap, 1, &tmpfmt);

	if (session->rtp) {
		struct ast_rtp_engine_ice *ice;

		ast_channel_set_fd(chan, 0, ast_rtp_instance_fd(session->rtp, 0));
		ast_channel_set_fd(chan, 1, ast_rtp_instance_fd(session->rtp, 1));
		ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(session->rtp), session->rtp, &session->prefs);

		if (((session->transport == JINGLE_TRANSPORT_GOOGLE_V2) ||
		     (session->transport == JINGLE_TRANSPORT_GOOGLE_V1)) &&
		    (ice = ast_rtp_instance_get_ice(session->rtp))) {
			/* We stop built in ICE support because we need to fall back to old old old STUN support */
			ice->stop(session->rtp);
		}
	}

	if (state == AST_STATE_RING) {
		ast_channel_rings_set(chan, 1);
	}

	ast_channel_adsicpe_set(chan, AST_ADSI_UNAVAILABLE);

	ast_best_codec(ast_channel_nativeformats(chan), &tmpfmt);
	ast_format_copy(ast_channel_writeformat(chan), &tmpfmt);
	ast_format_copy(ast_channel_rawwriteformat(chan), &tmpfmt);
	ast_format_copy(ast_channel_readformat(chan), &tmpfmt);
	ast_format_copy(ast_channel_rawreadformat(chan), &tmpfmt);

	ao2_lock(endpoint);

	ast_channel_callgroup_set(chan, endpoint->callgroup);
	ast_channel_pickupgroup_set(chan, endpoint->pickupgroup);

	if (!ast_strlen_zero(endpoint->accountcode)) {
		ast_channel_accountcode_set(chan, endpoint->accountcode);
	}

	if (!ast_strlen_zero(endpoint->language)) {
		ast_channel_language_set(chan, endpoint->language);
	}

	if (!ast_strlen_zero(endpoint->musicclass)) {
		ast_channel_musicclass_set(chan, endpoint->musicclass);
	}

	ast_channel_context_set(chan, endpoint->context);
	if (ast_exists_extension(NULL, endpoint->context, endpoint->name, 1, NULL)) {
		ast_channel_exten_set(chan, endpoint->name);
	} else {
		ast_channel_exten_set(chan, "s");
	}
	ast_channel_priority_set(chan, 1);

	ao2_unlock(endpoint);

	return chan;
}

/*! \brief Internal helper function which sends a response */
static void jingle_send_response(struct ast_xmpp_client *connection, ikspak *pak)
{
	iks *response;

	if (!(response = iks_new("iq"))) {
		ast_log(LOG_ERROR, "Unable to allocate an IKS response stanza\n");
		return;
	}

	iks_insert_attrib(response, "type", "result");
	iks_insert_attrib(response, "from", connection->jid->full);
	iks_insert_attrib(response, "to", iks_find_attrib(pak->x, "from"));
	iks_insert_attrib(response, "id", iks_find_attrib(pak->x, "id"));

	ast_xmpp_client_send(connection, response);

	iks_delete(response);
}

/*! \brief Internal helper function which sends an error response */
static void jingle_send_error_response(struct ast_xmpp_client *connection, ikspak *pak, const char *type, const char *reasonstr, const char *reasonstr2)
{
	iks *response, *error = NULL, *reason = NULL, *reason2 = NULL;

	if (!(response = iks_new("iq")) ||
	    !(error = iks_new("error")) ||
	    !(reason = iks_new(reasonstr))) {
		ast_log(LOG_ERROR, "Unable to allocate IKS error response stanzas\n");
		goto end;
	}

	iks_insert_attrib(response, "type", "error");
	iks_insert_attrib(response, "from", connection->jid->full);
	iks_insert_attrib(response, "to", iks_find_attrib(pak->x, "from"));
	iks_insert_attrib(response, "id", iks_find_attrib(pak->x, "id"));

	iks_insert_attrib(error, "type", type);
	iks_insert_node(error, reason);

	if (!ast_strlen_zero(reasonstr2) && (reason2 = iks_new(reasonstr2))) {
		iks_insert_node(error, reason2);
	}

	iks_insert_node(response, error);

	ast_xmpp_client_send(connection, response);
end:
	iks_delete(reason2);
	iks_delete(reason);
	iks_delete(error);
	iks_delete(response);
}

/*! \brief Internal helper function which adds ICE-UDP candidates to a transport node */
static int jingle_add_ice_udp_candidates_to_transport(struct ast_rtp_instance *rtp, iks *transport, iks **candidates, unsigned int maximum)
{
	struct ast_rtp_engine_ice *ice;
	struct ao2_container *local_candidates;
	struct ao2_iterator it;
	struct ast_rtp_engine_ice_candidate *candidate;
	int i = 0, res = 0;

	if (!(ice = ast_rtp_instance_get_ice(rtp)) || !(local_candidates = ice->get_local_candidates(rtp))) {
		ast_log(LOG_ERROR, "Unable to add ICE-UDP candidates as ICE support not available or no candidates available\n");
		return -1;
	}

	iks_insert_attrib(transport, "xmlns", JINGLE_ICE_UDP_NS);
	iks_insert_attrib(transport, "pwd", ice->get_password(rtp));
	iks_insert_attrib(transport, "ufrag", ice->get_ufrag(rtp));

	it = ao2_iterator_init(local_candidates, 0);

	while ((candidate = ao2_iterator_next(&it)) && (i < maximum)) {
		iks *local_candidate;
		char tmp[30];

		if (!(local_candidate = iks_new("candidate"))) {
			res = -1;
			ast_log(LOG_ERROR, "Unable to allocate IKS candidate stanza for ICE-UDP transport\n");
			break;
		}

		snprintf(tmp, sizeof(tmp), "%u", candidate->id);
		iks_insert_attrib(local_candidate, "component", tmp);
		snprintf(tmp, sizeof(tmp), "%d", ast_str_hash(candidate->foundation));
		iks_insert_attrib(local_candidate, "foundation", tmp);
		iks_insert_attrib(local_candidate, "generation", "0");
		iks_insert_attrib(local_candidate, "network", "0");
		snprintf(tmp, sizeof(tmp), "%04lx", (unsigned long)(ast_random() & 0xffff));
		iks_insert_attrib(local_candidate, "id", tmp);
		iks_insert_attrib(local_candidate, "ip", ast_sockaddr_stringify_host(&candidate->address));
		iks_insert_attrib(local_candidate, "port", ast_sockaddr_stringify_port(&candidate->address));
		snprintf(tmp, sizeof(tmp), "%d", candidate->priority);
		iks_insert_attrib(local_candidate, "priority", tmp);
		iks_insert_attrib(local_candidate, "protocol", "udp");

		if (candidate->type == AST_RTP_ICE_CANDIDATE_TYPE_HOST) {
			iks_insert_attrib(local_candidate, "type", "host");
		} else if (candidate->type == AST_RTP_ICE_CANDIDATE_TYPE_SRFLX) {
			iks_insert_attrib(local_candidate, "type", "srflx");
		} else if (candidate->type == AST_RTP_ICE_CANDIDATE_TYPE_RELAYED) {
			iks_insert_attrib(local_candidate, "type", "relay");
		}

		iks_insert_node(transport, local_candidate);
		candidates[i++] = local_candidate;
	}

	ao2_iterator_destroy(&it);
	ao2_ref(local_candidates, -1);

	return res;
}

/*! \brief Internal helper function which adds Google candidates to a transport node */
static int jingle_add_google_candidates_to_transport(struct ast_rtp_instance *rtp, iks *transport, iks **candidates, unsigned int video, enum jingle_transport transport_type, unsigned int maximum)
{
	struct ast_rtp_engine_ice *ice;
	struct ao2_container *local_candidates;
	struct ao2_iterator it;
	struct ast_rtp_engine_ice_candidate *candidate;
	int i = 0, res = 0;

	if (!(ice = ast_rtp_instance_get_ice(rtp)) || !(local_candidates = ice->get_local_candidates(rtp))) {
		ast_log(LOG_ERROR, "Unable to add Google ICE candidates as ICE support not available or no candidates available\n");
		return -1;
	}

	if (transport_type != JINGLE_TRANSPORT_GOOGLE_V1) {
		iks_insert_attrib(transport, "xmlns", GOOGLE_TRANSPORT_NS);
	}

	it = ao2_iterator_init(local_candidates, 0);

	while ((candidate = ao2_iterator_next(&it)) && (i < maximum)) {
		iks *local_candidate;
		/* In Google land a username is 16 bytes, explicitly */
		char ufrag[17] = "";

		if (!(local_candidate = iks_new("candidate"))) {
			res = -1;
			ast_log(LOG_ERROR, "Unable to allocate IKS candidate stanza for Google ICE transport\n");
			break;
		}

		if (candidate->id == 1) {
			iks_insert_attrib(local_candidate, "name", !video ? "rtp" : "video_rtp");
		} else if (candidate->id == 2) {
			iks_insert_attrib(local_candidate, "name", !video ? "rtcp" : "video_rtcp");
		} else {
			iks_delete(local_candidate);
			continue;
		}

		iks_insert_attrib(local_candidate, "address", ast_sockaddr_stringify_host(&candidate->address));
		iks_insert_attrib(local_candidate, "port", ast_sockaddr_stringify_port(&candidate->address));

		if (candidate->type == AST_RTP_ICE_CANDIDATE_TYPE_HOST) {
			iks_insert_attrib(local_candidate, "preference", "0.95");
			iks_insert_attrib(local_candidate, "type", "local");
		} else if (candidate->type == AST_RTP_ICE_CANDIDATE_TYPE_SRFLX) {
			iks_insert_attrib(local_candidate, "preference", "0.9");
			iks_insert_attrib(local_candidate, "type", "stun");
		}

		iks_insert_attrib(local_candidate, "protocol", "udp");
		iks_insert_attrib(local_candidate, "network", "0");
		snprintf(ufrag, sizeof(ufrag), "%s", ice->get_ufrag(rtp));
		iks_insert_attrib(local_candidate, "username", ufrag);
		iks_insert_attrib(local_candidate, "generation", "0");

		if (transport_type == JINGLE_TRANSPORT_GOOGLE_V1) {
			iks_insert_attrib(local_candidate, "password", "");
			iks_insert_attrib(local_candidate, "foundation", "0");
			iks_insert_attrib(local_candidate, "component", "1");
		} else {
			iks_insert_attrib(local_candidate, "password", ice->get_password(rtp));
		}

		/* You may notice a lack of relay support up above - this is because we don't support it for use with
		 * the Google talk transport due to their arcane support. */

		iks_insert_node(transport, local_candidate);
		candidates[i++] = local_candidate;
	}

	ao2_iterator_destroy(&it);
	ao2_ref(local_candidates, -1);

	return res;
}

/*! \brief Internal function which sends a session-terminate message */
static void jingle_send_session_terminate(struct jingle_session *session, const char *reasontext)
{
	iks *iq = NULL, *jingle = NULL, *reason = NULL, *text = NULL;

	if (!(iq = iks_new("iq")) || !(jingle = iks_new(session->transport == JINGLE_TRANSPORT_GOOGLE_V1 ? "session" : "jingle")) ||
	    !(reason = iks_new("reason")) || !(text = iks_new(reasontext))) {
		ast_log(LOG_ERROR, "Failed to allocate stanzas for session-terminate message on session '%s'\n", session->sid);
		goto end;
	}

	iks_insert_attrib(iq, "to", session->remote);
	iks_insert_attrib(iq, "type", "set");
	iks_insert_attrib(iq, "id", session->connection->mid);
	ast_xmpp_increment_mid(session->connection->mid);

	if (session->transport == JINGLE_TRANSPORT_GOOGLE_V1) {
		iks_insert_attrib(jingle, "type", "terminate");
		iks_insert_attrib(jingle, "id", session->sid);
		iks_insert_attrib(jingle, "xmlns", GOOGLE_SESSION_NS);
		iks_insert_attrib(jingle, "initiator", session->outgoing ? session->connection->jid->full : session->remote);
	} else {
		iks_insert_attrib(jingle, "action", "session-terminate");
		iks_insert_attrib(jingle, "sid", session->sid);
		iks_insert_attrib(jingle, "xmlns", JINGLE_NS);
	}

	iks_insert_node(iq, jingle);
	iks_insert_node(jingle, reason);
	iks_insert_node(reason, text);

	ast_xmpp_client_send(session->connection, iq);

end:
	iks_delete(text);
	iks_delete(reason);
	iks_delete(jingle);
	iks_delete(iq);
}

/*! \brief Internal function which sends a session-info message */
static void jingle_send_session_info(struct jingle_session *session, const char *info)
{
	iks *iq = NULL, *jingle = NULL, *text = NULL;

	/* Google-V1 has no way to send informational messages so don't even bother trying */
	if (session->transport == JINGLE_TRANSPORT_GOOGLE_V1) {
		return;
	}

	if (!(iq = iks_new("iq")) || !(jingle = iks_new("jingle")) || !(text = iks_new(info))) {
		ast_log(LOG_ERROR, "Failed to allocate stanzas for session-info message on session '%s'\n", session->sid);
		goto end;
	}

	iks_insert_attrib(iq, "to", session->remote);
	iks_insert_attrib(iq, "type", "set");
	iks_insert_attrib(iq, "id", session->connection->mid);
	ast_xmpp_increment_mid(session->connection->mid);

	iks_insert_attrib(jingle, "action", "session-info");
	iks_insert_attrib(jingle, "sid", session->sid);
	iks_insert_attrib(jingle, "xmlns", JINGLE_NS);
	iks_insert_node(iq, jingle);
	iks_insert_node(jingle, text);

	ast_xmpp_client_send(session->connection, iq);

end:
	iks_delete(text);
	iks_delete(jingle);
	iks_delete(iq);
}

/*! \internal
 *
 * \brief Locks both pvt and pvt owner if owner is present.
 *
 * \note This function gives a ref to pvt->owner if it is present and locked.
 *       This reference must be decremented after pvt->owner is unlocked.
 *
 * \note This function will never give you up,
 * \note This function will never let you down.
 * \note This function will run around and desert you.
 *
 * \pre pvt is not locked
 * \post pvt is locked
 * \post pvt->owner is locked and its reference count is increased (if pvt->owner is not NULL)
 *
 * \returns a pointer to the locked and reffed pvt->owner channel if it exists.
 */
static struct ast_channel *jingle_session_lock_full(struct jingle_session *pvt)
{
	struct ast_channel *chan;

	/* Locking is simple when it is done right.  If you see a deadlock resulting
	 * in this function, it is not this function's fault, Your problem exists elsewhere.
	 * This function is perfect... seriously. */
	for (;;) {
		/* First, get the channel and grab a reference to it */
		ao2_lock(pvt);
		chan = pvt->owner;
		if (chan) {
			/* The channel can not go away while we hold the pvt lock.
			 * Give the channel a ref so it will not go away after we let
			 * the pvt lock go. */
			ast_channel_ref(chan);
		} else {
			/* no channel, return pvt locked */
			return NULL;
		}

		/* We had to hold the pvt lock while getting a ref to the owner channel
		 * but now we have to let this lock go in order to preserve proper
		 * locking order when grabbing the channel lock */
		ao2_unlock(pvt);

		/* Look, no deadlock avoidance, hooray! */
		ast_channel_lock(chan);
		ao2_lock(pvt);
		if (pvt->owner == chan) {
			/* done */
			break;
		}

		/* If the owner changed while everything was unlocked, no problem,
		 * just start over and everthing will work.  This is rare, do not be
		 * confused by this loop and think this it is an expensive operation.
		 * The majority of the calls to this function will never involve multiple
		 * executions of this loop. */
		ast_channel_unlock(chan);
		ast_channel_unref(chan);
		ao2_unlock(pvt);
	}

	/* If owner exists, it is locked and reffed */
	return pvt->owner;
}

/*! \brief Helper function which queues a hangup frame with cause code */
static void jingle_queue_hangup_with_cause(struct jingle_session *session, int cause)
{
	struct ast_channel *chan;

	if ((chan = jingle_session_lock_full(session))) {
		ast_debug(3, "Hanging up channel '%s' with cause '%d'\n", ast_channel_name(chan), cause);
		ast_queue_hangup_with_cause(chan, cause);
		ast_channel_unlock(chan);
		ast_channel_unref(chan);
	}
	ao2_unlock(session);
}

/*! \brief Internal function which sends a transport-info message */
static void jingle_send_transport_info(struct jingle_session *session, const char *from)
{
	iks *iq, *jingle = NULL, *audio = NULL, *audio_transport = NULL, *video = NULL, *video_transport = NULL;
	iks *audio_candidates[session->maxicecandidates], *video_candidates[session->maxicecandidates];
	int i, res = 0;

	if (!(iq = iks_new("iq")) ||
	    !(jingle = iks_new(session->transport == JINGLE_TRANSPORT_GOOGLE_V1 ? "session" : "jingle"))) {
		iks_delete(iq);
		jingle_queue_hangup_with_cause(session, AST_CAUSE_SWITCH_CONGESTION);
		ast_log(LOG_ERROR, "Failed to allocate stanzas for transport-info message, hanging up session '%s'\n", session->sid);
		return;
	}

	memset(audio_candidates, 0, sizeof(audio_candidates));
	memset(video_candidates, 0, sizeof(video_candidates));

	iks_insert_attrib(iq, "from", session->connection->jid->full);
	iks_insert_attrib(iq, "to", from);
	iks_insert_attrib(iq, "type", "set");
	iks_insert_attrib(iq, "id", session->connection->mid);
	ast_xmpp_increment_mid(session->connection->mid);

	if (session->transport == JINGLE_TRANSPORT_GOOGLE_V1) {
		iks_insert_attrib(jingle, "type", "candidates");
		iks_insert_attrib(jingle, "id", session->sid);
		iks_insert_attrib(jingle, "xmlns", GOOGLE_SESSION_NS);
		iks_insert_attrib(jingle, "initiator", session->outgoing ? session->connection->jid->full : from);
	} else {
		iks_insert_attrib(jingle, "action", "transport-info");
		iks_insert_attrib(jingle, "sid", session->sid);
		iks_insert_attrib(jingle, "xmlns", JINGLE_NS);
	}
	iks_insert_node(iq, jingle);

	if (session->rtp) {
		if (session->transport == JINGLE_TRANSPORT_GOOGLE_V1) {
			/* V1 protocol has the candidates directly in the session */
			res = jingle_add_google_candidates_to_transport(session->rtp, jingle, audio_candidates, 0, session->transport, session->maxicecandidates);
		} else if ((audio = iks_new("content")) && (audio_transport = iks_new("transport"))) {
			iks_insert_attrib(audio, "creator", session->outgoing ? "initiator" : "responder");
			iks_insert_attrib(audio, "name", session->audio_name);
			iks_insert_node(jingle, audio);
			iks_insert_node(audio, audio_transport);

			if (session->transport == JINGLE_TRANSPORT_ICE_UDP) {
				res = jingle_add_ice_udp_candidates_to_transport(session->rtp, audio_transport, audio_candidates, session->maxicecandidates);
			} else if (session->transport == JINGLE_TRANSPORT_GOOGLE_V2) {
				res = jingle_add_google_candidates_to_transport(session->rtp, audio_transport, audio_candidates, 0, session->transport,
										session->maxicecandidates);
			}
		} else {
			res = -1;
		}
	}

	if ((session->transport != JINGLE_TRANSPORT_GOOGLE_V1) && !res && session->vrtp) {
		if ((video = iks_new("content")) && (video_transport = iks_new("transport"))) {
			iks_insert_attrib(video, "creator", session->outgoing ? "initiator" : "responder");
			iks_insert_attrib(video, "name", session->video_name);
			iks_insert_node(jingle, video);
			iks_insert_node(video, video_transport);

			if (session->transport == JINGLE_TRANSPORT_ICE_UDP) {
				res = jingle_add_ice_udp_candidates_to_transport(session->vrtp, video_transport, video_candidates, session->maxicecandidates);
			} else if (session->transport == JINGLE_TRANSPORT_GOOGLE_V2) {
				res = jingle_add_google_candidates_to_transport(session->vrtp, video_transport, video_candidates, 1, session->transport,
										session->maxicecandidates);
			}
		} else {
			res = -1;
		}
	}

	if (!res) {
		ast_xmpp_client_send(session->connection, iq);
	} else {
		jingle_queue_hangup_with_cause(session, AST_CAUSE_SWITCH_CONGESTION);
	}

	/* Clean up after ourselves */
	for (i = 0; i < session->maxicecandidates; i++) {
		iks_delete(video_candidates[i]);
		iks_delete(audio_candidates[i]);
	}

	iks_delete(video_transport);
	iks_delete(video);
	iks_delete(audio_transport);
	iks_delete(audio);
	iks_delete(jingle);
	iks_delete(iq);
}

/*! \brief Internal helper function which adds payloads to a description */
static int jingle_add_payloads_to_description(struct jingle_session *session, struct ast_rtp_instance *rtp, iks *description, iks **payloads, enum ast_format_type type)
{
	struct ast_format format;
	int x = 0, i = 0, res = 0;

	for (x = 0; (x < AST_CODEC_PREF_SIZE) && (i < (session->maxpayloads - 2)); x++) {
		int rtp_code;
		iks *payload;
		char tmp[32];

		if (!ast_codec_pref_index(&session->prefs, x, &format)) {
			break;
		}

		if (AST_FORMAT_GET_TYPE(format.id) != type) {
			continue;
		}

		if (!ast_format_cap_iscompatible(session->jointcap, &format)) {
			continue;
		}

		if (((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(rtp), 1, &format, 0)) == -1) ||
		    (!(payload = iks_new("payload-type")))) {
			return -1;
		}

		if (session->transport == JINGLE_TRANSPORT_GOOGLE_V1) {
			iks_insert_attrib(payload, "xmlns", GOOGLE_PHONE_NS);
		}

		snprintf(tmp, sizeof(tmp), "%d", rtp_code);
		iks_insert_attrib(payload, "id", tmp);
		iks_insert_attrib(payload, "name", ast_rtp_lookup_mime_subtype2(1, &format, 0, 0));
		iks_insert_attrib(payload, "channels", "1");

		if ((format.id == AST_FORMAT_G722) && ((session->transport == JINGLE_TRANSPORT_GOOGLE_V1) || (session->transport == JINGLE_TRANSPORT_GOOGLE_V2))) {
			iks_insert_attrib(payload, "clockrate", "16000");
		} else {
			snprintf(tmp, sizeof(tmp), "%u", ast_rtp_lookup_sample_rate2(1, &format, 0));
			iks_insert_attrib(payload, "clockrate", tmp);
		}

		if ((type == AST_FORMAT_TYPE_VIDEO) && (session->transport == JINGLE_TRANSPORT_GOOGLE_V2)) {
			iks *parameter;

			/* Google requires these parameters to be set, but alas we can not give accurate values so use some safe defaults */
			if ((parameter = iks_new("parameter"))) {
				iks_insert_attrib(parameter, "name", "width");
				iks_insert_attrib(parameter, "value", "640");
				iks_insert_node(payload, parameter);
			}
			if ((parameter = iks_new("parameter"))) {
				iks_insert_attrib(parameter, "name", "height");
				iks_insert_attrib(parameter, "value", "480");
				iks_insert_node(payload, parameter);
			}
			if ((parameter = iks_new("parameter"))) {
				iks_insert_attrib(parameter, "name", "framerate");
				iks_insert_attrib(parameter, "value", "30");
				iks_insert_node(payload, parameter);
			}
		}

		iks_insert_node(description, payload);
		payloads[i++] = payload;
	}
	/* If this is for audio and there is room for RFC2833 add it in */
	if ((type == AST_FORMAT_TYPE_AUDIO) && (i < session->maxpayloads)) {
		iks *payload;

		if ((payload = iks_new("payload-type"))) {
			if (session->transport == JINGLE_TRANSPORT_GOOGLE_V1) {
				iks_insert_attrib(payload, "xmlns", GOOGLE_PHONE_NS);
			}

			iks_insert_attrib(payload, "id", "101");
			iks_insert_attrib(payload, "name", "telephone-event");
			iks_insert_attrib(payload, "channels", "1");
			iks_insert_attrib(payload, "clockrate", "8000");
			iks_insert_node(description, payload);
			payloads[i++] = payload;
		}
	}

	return res;
}

/*! \brief Helper function which adds content to a description */
static int jingle_add_content(struct jingle_session *session, iks *jingle, iks *content, iks *description, iks *transport,
			      const char *name, enum ast_format_type type, struct ast_rtp_instance *rtp, iks **payloads)
{
	int res = 0;

	if (session->transport != JINGLE_TRANSPORT_GOOGLE_V1) {
		iks_insert_attrib(content, "creator", session->outgoing ? "initiator" : "responder");
		iks_insert_attrib(content, "name", name);
		iks_insert_node(jingle, content);

		iks_insert_attrib(description, "xmlns", JINGLE_RTP_NS);
		if (type == AST_FORMAT_TYPE_AUDIO) {
			iks_insert_attrib(description, "media", "audio");
		} else if (type == AST_FORMAT_TYPE_VIDEO) {
			iks_insert_attrib(description, "media", "video");
		} else {
			return -1;
		}
		iks_insert_node(content, description);
	} else {
		iks_insert_attrib(description, "xmlns", GOOGLE_PHONE_NS);
		iks_insert_node(jingle, description);
	}

	if (!(res = jingle_add_payloads_to_description(session, rtp, description, payloads, type))) {
		if (session->transport == JINGLE_TRANSPORT_ICE_UDP) {
			iks_insert_attrib(transport, "xmlns", JINGLE_ICE_UDP_NS);
			iks_insert_node(content, transport);
		} else if (session->transport == JINGLE_TRANSPORT_GOOGLE_V2) {
			iks_insert_attrib(transport, "xmlns", GOOGLE_TRANSPORT_NS);
			iks_insert_node(content, transport);
		}
	}

	return res;
}

/*! \brief Internal function which sends a complete session message */
static void jingle_send_session_action(struct jingle_session *session, const char *action)
{
	iks *iq, *jingle, *audio = NULL, *audio_description = NULL, *video = NULL, *video_description = NULL;
	iks *audio_payloads[session->maxpayloads], *video_payloads[session->maxpayloads];
	iks *audio_transport = NULL, *video_transport = NULL;
	int i, res = 0;

	if (!(iq = iks_new("iq")) ||
	    !(jingle = iks_new(session->transport == JINGLE_TRANSPORT_GOOGLE_V1 ? "session" : "jingle"))) {
		jingle_queue_hangup_with_cause(session, AST_CAUSE_SWITCH_CONGESTION);
		iks_delete(iq);
		return;
	}

	memset(audio_payloads, 0, sizeof(audio_payloads));
	memset(video_payloads, 0, sizeof(video_payloads));

	iks_insert_attrib(iq, "from", session->connection->jid->full);
	iks_insert_attrib(iq, "to", session->remote);
	iks_insert_attrib(iq, "type", "set");
	iks_insert_attrib(iq, "id", session->connection->mid);
	ast_xmpp_increment_mid(session->connection->mid);

	if (session->transport == JINGLE_TRANSPORT_GOOGLE_V1) {
		iks_insert_attrib(jingle, "type", action);
		iks_insert_attrib(jingle, "id", session->sid);
		iks_insert_attrib(jingle, "xmlns", GOOGLE_SESSION_NS);
	} else {
		iks_insert_attrib(jingle, "action", action);
		iks_insert_attrib(jingle, "sid", session->sid);
		iks_insert_attrib(jingle, "xmlns", JINGLE_NS);
	}

	if (!strcasecmp(action, "session-initiate") || !strcasecmp(action, "initiate") || !strcasecmp(action, "accept")) {
		iks_insert_attrib(jingle, "initiator", session->outgoing ? session->connection->jid->full : session->remote);
	}

	iks_insert_node(iq, jingle);

	if (session->rtp && (audio = iks_new("content")) && (audio_description = iks_new("description")) &&
	    (audio_transport = iks_new("transport"))) {
		res = jingle_add_content(session, jingle, audio, audio_description, audio_transport, session->audio_name,
					 AST_FORMAT_TYPE_AUDIO, session->rtp, audio_payloads);
	} else {
		ast_log(LOG_ERROR, "Failed to allocate audio content stanzas for session '%s', hanging up\n", session->sid);
		res = -1;
	}

	if ((session->transport != JINGLE_TRANSPORT_GOOGLE_V1) && !res && session->vrtp) {
		if ((video = iks_new("content")) && (video_description = iks_new("description")) &&
		    (video_transport = iks_new("transport"))) {
			res = jingle_add_content(session, jingle, video, video_description, video_transport, session->video_name,
						 AST_FORMAT_TYPE_VIDEO, session->vrtp, video_payloads);
		} else {
			ast_log(LOG_ERROR, "Failed to allocate video content stanzas for session '%s', hanging up\n", session->sid);
			res = -1;
		}
	}

	if (!res) {
		ast_xmpp_client_send(session->connection, iq);
	} else {
		jingle_queue_hangup_with_cause(session, AST_CAUSE_SWITCH_CONGESTION);
	}

	iks_delete(video_transport);
	iks_delete(audio_transport);

	for (i = 0; i < session->maxpayloads; i++) {
		iks_delete(video_payloads[i]);
		iks_delete(audio_payloads[i]);
	}

	iks_delete(video_description);
	iks_delete(video);
	iks_delete(audio_description);
	iks_delete(audio);
	iks_delete(jingle);
	iks_delete(iq);
}

/*! \brief Internal function which sends a session-inititate message */
static void jingle_send_session_initiate(struct jingle_session *session)
{
	jingle_send_session_action(session, session->transport == JINGLE_TRANSPORT_GOOGLE_V1 ? "initiate" : "session-initiate");
}

/*! \brief Internal function which sends a session-accept message */
static void jingle_send_session_accept(struct jingle_session *session)
{
	jingle_send_session_action(session, session->transport == JINGLE_TRANSPORT_GOOGLE_V1 ? "accept" : "session-accept");
}

/*! \brief Callback for when a response is received for an outgoing session-initiate message */
static int jingle_outgoing_hook(void *data, ikspak *pak)
{
	struct jingle_session *session = data;
	iks *error = iks_find(pak->x, "error"), *redirect;

	/* In all cases this hook is done with */
	iks_filter_remove_rule(session->connection->filter, session->rule);
	session->rule = NULL;

	ast_callid_threadassoc_add(session->callid);

	/* If no error occurred they accepted our session-initiate message happily */
	if (!error) {
		struct ast_channel *chan;

		if ((chan = jingle_session_lock_full(session))) {
			ast_queue_control(chan, AST_CONTROL_PROCEEDING);
			ast_channel_unlock(chan);
			ast_channel_unref(chan);
		}
		ao2_unlock(session);

		jingle_send_transport_info(session, iks_find_attrib(pak->x, "from"));

		goto end;
	}

	/* Assume that because this is an error the session is gone, there is only one case where this is incorrect - a redirect */
	session->gone = 1;

	/* Map the error we received to an appropriate cause code and hang up the channel */
	if ((redirect = iks_find_with_attrib(error, "redirect", "xmlns", XMPP_STANZAS_NS))) {
		iks *to = iks_child(redirect);
		char *target;

		if (to && (target = iks_name(to)) && !ast_strlen_zero(target)) {
			/* Make the xmpp: go away if it is present */
			if (!strncmp(target, "xmpp:", 5)) {
				target += 5;
			}

			/* This is actually a fairly simple operation - we update the remote and send another session-initiate */
			ast_copy_string(session->remote, target, sizeof(session->remote));

			/* Add a new hook so we can get the status of redirected session */
			session->rule = iks_filter_add_rule(session->connection->filter, jingle_outgoing_hook, session,
							    IKS_RULE_ID, session->connection->mid, IKS_RULE_DONE);

			jingle_send_session_initiate(session);

			session->gone = 0;
		} else {
			jingle_queue_hangup_with_cause(session, AST_CAUSE_PROTOCOL_ERROR);
		}
	} else if (iks_find_with_attrib(error, "service-unavailable", "xmlns", XMPP_STANZAS_NS)) {
		jingle_queue_hangup_with_cause(session, AST_CAUSE_CONGESTION);
	} else if (iks_find_with_attrib(error, "resource-constraint", "xmlns", XMPP_STANZAS_NS)) {
		jingle_queue_hangup_with_cause(session, AST_CAUSE_REQUESTED_CHAN_UNAVAIL);
	} else if (iks_find_with_attrib(error, "bad-request", "xmlns", XMPP_STANZAS_NS)) {
		jingle_queue_hangup_with_cause(session, AST_CAUSE_PROTOCOL_ERROR);
	} else if (iks_find_with_attrib(error, "remote-server-not-found", "xmlns", XMPP_STANZAS_NS)) {
		jingle_queue_hangup_with_cause(session, AST_CAUSE_NO_ROUTE_DESTINATION);
	} else if (iks_find_with_attrib(error, "feature-not-implemented", "xmlns", XMPP_STANZAS_NS)) {
		/* Assume that this occurred because the remote side does not support our transport, so drop it down one and try again */
		session->transport--;

		/* If we still have a viable transport mechanism re-send the session-initiate */
		if (session->transport != JINGLE_TRANSPORT_NONE) {
			struct ast_rtp_engine_ice *ice;

			if (((session->transport == JINGLE_TRANSPORT_GOOGLE_V2) ||
			     (session->transport == JINGLE_TRANSPORT_GOOGLE_V1)) &&
			    (ice = ast_rtp_instance_get_ice(session->rtp))) {
				/* We stop built in ICE support because we need to fall back to old old old STUN support */
				ice->stop(session->rtp);
			}

			/* Re-send the message to the *original* target and not a redirected one */
			ast_copy_string(session->remote, session->remote_original, sizeof(session->remote));

			session->rule = iks_filter_add_rule(session->connection->filter, jingle_outgoing_hook, session,
							    IKS_RULE_ID, session->connection->mid, IKS_RULE_DONE);

			jingle_send_session_initiate(session);

			session->gone = 0;
		} else {
			/* Otherwise we have exhausted all transports */
			jingle_queue_hangup_with_cause(session, AST_CAUSE_FACILITY_NOT_IMPLEMENTED);
		}
	} else {
		jingle_queue_hangup_with_cause(session, AST_CAUSE_PROTOCOL_ERROR);
	}

end:
	ast_callid_threadassoc_remove();

	return IKS_FILTER_EAT;
}

/*! \brief Function called by core when we should answer a Jingle session */
static int jingle_answer(struct ast_channel *ast)
{
	struct jingle_session *session = ast_channel_tech_pvt(ast);

	/* The channel has already been answered so we don't need to do anything */
	if (ast_channel_state(ast) == AST_STATE_UP) {
		return 0;
	}

	jingle_send_session_accept(session);

	return 0;
}

/*! \brief Function called by core to read any waiting frames */
static struct ast_frame *jingle_read(struct ast_channel *ast)
{
	struct jingle_session *session = ast_channel_tech_pvt(ast);
	struct ast_frame *frame = &ast_null_frame;

	switch (ast_channel_fdno(ast)) {
	case 0:
		if (session->rtp) {
			frame = ast_rtp_instance_read(session->rtp, 0);
		}
		break;
	case 1:
		if (session->rtp) {
			frame = ast_rtp_instance_read(session->rtp, 1);
		}
		break;
	case 2:
		if (session->vrtp) {
			frame = ast_rtp_instance_read(session->vrtp, 0);
		}
		break;
	case 3:
		if (session->vrtp) {
			frame = ast_rtp_instance_read(session->vrtp, 1);
		}
		break;
	default:
		break;
	}

	if (frame && frame->frametype == AST_FRAME_VOICE &&
	    !ast_format_cap_iscompatible(ast_channel_nativeformats(ast), &frame->subclass.format)) {
		if (!ast_format_cap_iscompatible(session->jointcap, &frame->subclass.format)) {
			ast_debug(1, "Bogus frame of format '%s' received from '%s'!\n",
				  ast_getformatname(&frame->subclass.format), ast_channel_name(ast));
			ast_frfree(frame);
			frame = &ast_null_frame;
		} else {
			ast_debug(1, "Oooh, format changed to %s\n",
				  ast_getformatname(&frame->subclass.format));
			ast_format_cap_remove_bytype(ast_channel_nativeformats(ast), AST_FORMAT_TYPE_AUDIO);
			ast_format_cap_add(ast_channel_nativeformats(ast), &frame->subclass.format);
			ast_set_read_format(ast, ast_channel_readformat(ast));
			ast_set_write_format(ast, ast_channel_writeformat(ast));
		}
	}

	return frame;
}

/*! \brief Function called by core to write frames */
static int jingle_write(struct ast_channel *ast, struct ast_frame *frame)
{
	struct jingle_session *session = ast_channel_tech_pvt(ast);
	int res = 0;
	char buf[256];

	switch (frame->frametype) {
	case AST_FRAME_VOICE:
		if (!(ast_format_cap_iscompatible(ast_channel_nativeformats(ast), &frame->subclass.format))) {
			ast_log(LOG_WARNING,
				"Asked to transmit frame type %s, while native formats is %s (read/write = %s/%s)\n",
				ast_getformatname(&frame->subclass.format),
				ast_getformatname_multiple(buf, sizeof(buf), ast_channel_nativeformats(ast)),
				ast_getformatname(ast_channel_readformat(ast)),
				ast_getformatname(ast_channel_writeformat(ast)));
			return 0;
		}
		if (session && session->rtp) {
			res = ast_rtp_instance_write(session->rtp, frame);
		}
		break;
	case AST_FRAME_VIDEO:
		if (session && session->vrtp) {
			res = ast_rtp_instance_write(session->vrtp, frame);
		}
		break;
	default:
		ast_log(LOG_WARNING, "Can't send %u type frames with Jingle write\n",
			frame->frametype);
		return 0;
	}

	return res;
}

/*! \brief Function called by core to change the underlying owner channel */
static int jingle_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
{
	struct jingle_session *session = ast_channel_tech_pvt(newchan);

	ao2_lock(session);

	session->owner = newchan;

	ao2_unlock(session);

	return 0;
}

/*! \brief Function called by core to ask the channel to indicate some sort of condition */
static int jingle_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen)
{
	struct jingle_session *session = ast_channel_tech_pvt(ast);
	int res = 0;

	switch (condition) {
	case AST_CONTROL_RINGING:
		if (ast_channel_state(ast) == AST_STATE_RING) {
			jingle_send_session_info(session, "ringing xmlns='urn:xmpp:jingle:apps:rtp:info:1'");
		} else {
			res = -1;
		}
		break;
	case AST_CONTROL_BUSY:
		if (ast_channel_state(ast) != AST_STATE_UP) {
			ast_channel_hangupcause_set(ast, AST_CAUSE_BUSY);
			ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
		} else {
			res = -1;
		}
		break;
	case AST_CONTROL_CONGESTION:
		if (ast_channel_state(ast) != AST_STATE_UP) {
			ast_channel_hangupcause_set(ast, AST_CAUSE_CONGESTION);
			ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
		} else {
			res = -1;
		}
		break;
	case AST_CONTROL_INCOMPLETE:
		if (ast_channel_state(ast) != AST_STATE_UP) {
			ast_channel_hangupcause_set(ast, AST_CAUSE_CONGESTION);
			ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
		}
		break;
	case AST_CONTROL_HOLD:
		ast_moh_start(ast, data, NULL);
		break;
	case AST_CONTROL_UNHOLD:
		ast_moh_stop(ast);
		break;
	case AST_CONTROL_SRCUPDATE:
		if (session->rtp) {
			ast_rtp_instance_update_source(session->rtp);
		}
		break;
	case AST_CONTROL_SRCCHANGE:
		if (session->rtp) {
			ast_rtp_instance_change_source(session->rtp);
		}
		break;
	case AST_CONTROL_VIDUPDATE:
	case AST_CONTROL_UPDATE_RTP_PEER:
	case AST_CONTROL_CONNECTED_LINE:
		break;
	case AST_CONTROL_PVT_CAUSE_CODE:
	case -1:
		res = -1;
		break;
	default:
		ast_log(LOG_NOTICE, "Don't know how to indicate condition '%d'\n", condition);
		res = -1;
	}

	return res;
}

/*! \brief Function called by core to send text to the remote party of the Jingle session */
static int jingle_sendtext(struct ast_channel *chan, const char *text)
{
	struct jingle_session *session = ast_channel_tech_pvt(chan);

	return ast_xmpp_client_send_message(session->connection, session->remote, text);
}

/*! \brief Function called by core to start a DTMF digit */
static int jingle_digit_begin(struct ast_channel *chan, char digit)
{
	struct jingle_session *session = ast_channel_tech_pvt(chan);

	if (session->rtp) {
		ast_rtp_instance_dtmf_begin(session->rtp, digit);
	}

	return 0;
}

/*! \brief Function called by core to stop a DTMF digit */
static int jingle_digit_end(struct ast_channel *ast, char digit, unsigned int duration)
{
	struct jingle_session *session = ast_channel_tech_pvt(ast);

	if (session->rtp) {
		ast_rtp_instance_dtmf_end_with_duration(session->rtp, digit, duration);
	}

	return 0;
}

/*! \brief Function called by core to actually start calling a remote party */
static int jingle_call(struct ast_channel *ast, const char *dest, int timeout)
{
	struct jingle_session *session = ast_channel_tech_pvt(ast);

	ast_setstate(ast, AST_STATE_RING);

	/* Since we have no idea of the remote capabilities use ours for now */
	ast_format_cap_copy(session->jointcap, session->cap);

	/* We set up a hook so we can know when our session-initiate message was accepted or rejected */
	session->rule = iks_filter_add_rule(session->connection->filter, jingle_outgoing_hook, session,
					    IKS_RULE_ID, session->connection->mid, IKS_RULE_DONE);

	jingle_send_session_initiate(session);

	return 0;
}

/*! \brief Function called by core to hang up a Jingle session */
static int jingle_hangup(struct ast_channel *ast)
{
	struct jingle_session *session = ast_channel_tech_pvt(ast);

	ao2_lock(session);

	if ((ast_channel_state(ast) != AST_STATE_DOWN) && !session->gone) {
		int cause = (session->owner ? ast_channel_hangupcause(session->owner) : AST_CAUSE_CONGESTION);
		const char *reason = "success";
		int i;

		/* Get the appropriate reason and send a session-terminate */
		for (i = 0; i < ARRAY_LEN(jingle_reason_mappings); i++) {
			if (jingle_reason_mappings[i].cause == cause) {
				reason = jingle_reason_mappings[i].reason;
				break;
			}
		}

		jingle_send_session_terminate(session, reason);
	}

	ast_channel_tech_pvt_set(ast, NULL);
	session->owner = NULL;

	ao2_unlink(session->state->sessions, session);
	ao2_ref(session->state, -1);

	ao2_unlock(session);
	ao2_ref(session, -1);

	return 0;
}

/*! \brief Function called by core to create a new outgoing Jingle session */
static struct ast_channel *jingle_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, const char *data, int *cause)
{
	RAII_VAR(struct jingle_config *, cfg, ao2_global_obj_ref(globals), ao2_cleanup);
	RAII_VAR(struct jingle_endpoint *, endpoint, NULL, ao2_cleanup);
	char *dialed, target[200] = "";
	struct ast_xmpp_buddy *buddy;
	struct jingle_session *session;
	struct ast_channel *chan;
	enum jingle_transport transport = JINGLE_TRANSPORT_NONE;
	struct ast_rtp_engine_ice *ice;
	AST_DECLARE_APP_ARGS(args,
			     AST_APP_ARG(name);
			     AST_APP_ARG(target);
		);

	/* We require at a minimum one audio format to be requested */
	if (!ast_format_cap_has_type(cap, AST_FORMAT_TYPE_AUDIO)) {
		ast_log(LOG_ERROR, "Motif channel driver requires an audio format when dialing a destination\n");
		*cause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
		return NULL;
	}

	if (ast_strlen_zero(data) || !(dialed = ast_strdupa(data))) {
		ast_log(LOG_ERROR, "Unable to create channel with empty destination.\n");
		*cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
		return NULL;
	}

	/* Parse the given dial string and validate the results */
	AST_NONSTANDARD_APP_ARGS(args, dialed, '/');

	if (ast_strlen_zero(args.name) || ast_strlen_zero(args.target)) {
		ast_log(LOG_ERROR, "Unable to determine endpoint name and target.\n");
		*cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
		return NULL;
	}

	if (!(endpoint = jingle_endpoint_find(cfg->endpoints, args.name))) {
		ast_log(LOG_ERROR, "Endpoint '%s' does not exist.\n", args.name);
		*cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
		return NULL;
	}

	ao2_lock(endpoint->state);

	/* If we don't have a connection for the endpoint we can't exactly start a session on it */
	if (!endpoint->connection) {
		ast_log(LOG_ERROR, "Unable to create Jingle session on endpoint '%s' as no valid connection exists\n", args.name);
		*cause = AST_CAUSE_SWITCH_CONGESTION;
		ao2_unlock(endpoint->state);
		return NULL;
	}

	/* Find the target in the roster so we can choose a resource */
	if ((buddy = ao2_find(endpoint->connection->buddies, args.target, OBJ_KEY))) {
		struct ao2_iterator res;
		struct ast_xmpp_resource *resource;

		/* Iterate through finding the first viable Jingle capable resource */
		res = ao2_iterator_init(buddy->resources, 0);
		while ((resource = ao2_iterator_next(&res))) {
			if (resource->caps.jingle) {
				snprintf(target, sizeof(target), "%s/%s", args.target, resource->resource);
				transport = JINGLE_TRANSPORT_ICE_UDP;
				break;
			} else if (resource->caps.google) {
				snprintf(target, sizeof(target), "%s/%s", args.target, resource->resource);
				transport = JINGLE_TRANSPORT_GOOGLE_V2;
				break;
			}
			ao2_ref(resource, -1);
		}
		ao2_iterator_destroy(&res);

		ao2_ref(buddy, -1);
	} else {
		/* If the target is NOT in the roster use the provided target as-is */
		ast_copy_string(target, args.target, sizeof(target));
	}

	ao2_unlock(endpoint->state);

	/* If no target was found we can't set up a session */
	if (ast_strlen_zero(target)) {
		ast_log(LOG_ERROR, "Unable to create Jingle session on endpoint '%s' as no capable resource for target '%s' was found\n", args.name, args.target);
		*cause = AST_CAUSE_SWITCH_CONGESTION;
		return NULL;
	}

	if (!(session = jingle_alloc(endpoint, target, NULL))) {
		ast_log(LOG_ERROR, "Unable to create Jingle session on endpoint '%s'\n", args.name);
		*cause = AST_CAUSE_SWITCH_CONGESTION;
		return NULL;
	}

	/* Update the transport if we learned what we should actually use */
	if (transport != JINGLE_TRANSPORT_NONE) {
		session->transport = transport;
		/* Note that for Google-V1 and Google-V2 we don't stop built-in ICE support, this will happen in jingle_new */
	}

	if (!(chan = jingle_new(endpoint, session, AST_STATE_DOWN, target, requestor ? ast_channel_linkedid(requestor) : NULL, NULL))) {
		ast_log(LOG_ERROR, "Unable to create Jingle channel on endpoint '%s'\n", args.name);
		*cause = AST_CAUSE_SWITCH_CONGESTION;
		ao2_ref(session, -1);
		return NULL;
	}

	/* If video was requested try to enable it on the session */
	if (ast_format_cap_has_type(cap, AST_FORMAT_TYPE_VIDEO)) {
		jingle_enable_video(session);
	}

	/* As this is outgoing set ourselves as controlling */
	if (session->rtp && (ice = ast_rtp_instance_get_ice(session->rtp))) {
		ice->ice_lite(session->rtp);
	}

	if (session->vrtp && (ice = ast_rtp_instance_get_ice(session->vrtp))) {
		ice->ice_lite(session->vrtp);
	}

	/* We purposely don't decrement the session here as there is a reference on the channel */
	ao2_link(endpoint->state->sessions, session);

	return chan;
}

/*! \brief Helper function which handles content descriptions */
static int jingle_interpret_description(struct jingle_session *session, iks *description, const char *name, struct ast_rtp_instance **rtp)
{
	char *media = iks_find_attrib(description, "media");
	struct ast_rtp_codecs codecs;
	iks *codec;
	int othercapability = 0;

	/* Google-V1 is always carrying audio, but just doesn't tell us so */
	if (session->transport == JINGLE_TRANSPORT_GOOGLE_V1) {
		media = "audio";
	} else if (ast_strlen_zero(media)) {
		jingle_queue_hangup_with_cause(session, AST_CAUSE_BEARERCAPABILITY_NOTAVAIL);
		ast_log(LOG_ERROR, "Received a content description on session '%s' without a name\n", session->sid);
		return -1;
	}

	/* Determine the type of media that is being carried and update the RTP instance, as well as the name */
	if (!strcasecmp(media, "audio")) {
		if (!ast_strlen_zero(name)) {
			ast_string_field_set(session, audio_name, name);
		}
		*rtp = session->rtp;
		ast_format_cap_remove_bytype(session->peercap, AST_FORMAT_TYPE_AUDIO);
		ast_format_cap_remove_bytype(session->jointcap, AST_FORMAT_TYPE_AUDIO);
	} else if (!strcasecmp(media, "video")) {
		if (!ast_strlen_zero(name)) {
			ast_string_field_set(session, video_name, name);
		}

		jingle_enable_video(session);
		*rtp = session->vrtp;

		/* If video is not present cancel this session */
		if (!session->vrtp) {
			jingle_queue_hangup_with_cause(session, AST_CAUSE_BEARERCAPABILITY_NOTAVAIL);
			ast_log(LOG_ERROR, "Received a video content description on session '%s' but could not enable video\n", session->sid);
			return -1;
		}

		ast_format_cap_remove_bytype(session->peercap, AST_FORMAT_TYPE_VIDEO);
		ast_format_cap_remove_bytype(session->jointcap, AST_FORMAT_TYPE_VIDEO);
	} else {
		/* Unknown media type */
		jingle_queue_hangup_with_cause(session, AST_CAUSE_BEARERCAPABILITY_NOTAVAIL);
		ast_log(LOG_ERROR, "Unsupported media type '%s' received in content description on session '%s'\n", media, session->sid);
		return -1;
	}

	if (ast_rtp_codecs_payloads_initialize(&codecs)) {
		jingle_queue_hangup_with_cause(session, AST_CAUSE_BEARERCAPABILITY_NOTAVAIL);
		ast_log(LOG_ERROR, "Could not initialize codecs for negotiation on session '%s'\n", session->sid);
		return -1;
	}

	/* Iterate the codecs updating the relevant RTP instance as we go */
	for (codec = iks_child(description); codec; codec = iks_next(codec)) {
		char *id = iks_find_attrib(codec, "id"), *name = iks_find_attrib(codec, "name");
		char *clockrate = iks_find_attrib(codec, "clockrate");
		int rtp_id, rtp_clockrate;

		if (!ast_strlen_zero(id) && !ast_strlen_zero(name) && (sscanf(id, "%30d", &rtp_id) == 1)) {
			ast_rtp_codecs_payloads_set_m_type(&codecs, NULL, rtp_id);

			if (!ast_strlen_zero(clockrate) && (sscanf(clockrate, "%30d", &rtp_clockrate) == 1)) {
				ast_rtp_codecs_payloads_set_rtpmap_type_rate(&codecs, NULL, rtp_id, media, name, 0, rtp_clockrate);
			} else {
				ast_rtp_codecs_payloads_set_rtpmap_type(&codecs, NULL, rtp_id, media, name, 0);
			}
		}
	}

	ast_rtp_codecs_payload_formats(&codecs, session->peercap, &othercapability);
	ast_format_cap_joint_append(session->cap, session->peercap, session->jointcap);

	if (ast_format_cap_is_empty(session->jointcap)) {
		/* We have no compatible codecs, so terminate the session appropriately */
		jingle_queue_hangup_with_cause(session, AST_CAUSE_BEARERCAPABILITY_NOTAVAIL);
		ast_rtp_codecs_payloads_destroy(&codecs);
		return -1;
	}

	ast_rtp_codecs_payloads_copy(&codecs, ast_rtp_instance_get_codecs(*rtp), *rtp);
	ast_rtp_codecs_payloads_destroy(&codecs);

	return 0;
}

/*! \brief Helper function which handles ICE-UDP transport information */
static int jingle_interpret_ice_udp_transport(struct jingle_session *session, iks *transport, struct ast_rtp_instance *rtp)
{
	struct ast_rtp_engine_ice *ice = ast_rtp_instance_get_ice(rtp);
	char *ufrag = iks_find_attrib(transport, "ufrag"), *pwd = iks_find_attrib(transport, "pwd");
	iks *candidate;

	if (!ice) {
		jingle_queue_hangup_with_cause(session, AST_CAUSE_SWITCH_CONGESTION);
		ast_log(LOG_ERROR, "Received ICE-UDP transport information on session '%s' but ICE support not available\n", session->sid);
		return -1;
	}

	if (!ast_strlen_zero(ufrag) && !ast_strlen_zero(pwd)) {
		ice->set_authentication(rtp, ufrag, pwd);
	}

	for (candidate = iks_child(transport); candidate; candidate = iks_next(candidate)) {
		char *component = iks_find_attrib(candidate, "component"), *foundation = iks_find_attrib(candidate, "foundation");
		char *generation = iks_find_attrib(candidate, "generation"), *id = iks_find_attrib(candidate, "id");
		char *ip = iks_find_attrib(candidate, "ip"), *port = iks_find_attrib(candidate, "port");
		char *priority = iks_find_attrib(candidate, "priority"), *protocol = iks_find_attrib(candidate, "protocol");
		char *type = iks_find_attrib(candidate, "type");
		struct ast_rtp_engine_ice_candidate local_candidate = { 0, };
		int real_port;
		struct ast_sockaddr remote_address = { { 0, } };

		/* If this candidate is incomplete skip it */
		if (ast_strlen_zero(component) || ast_strlen_zero(foundation) || ast_strlen_zero(generation) || ast_strlen_zero(id) ||
		    ast_strlen_zero(ip) || ast_strlen_zero(port) || ast_strlen_zero(priority) ||
		    ast_strlen_zero(protocol) || ast_strlen_zero(type)) {
			jingle_queue_hangup_with_cause(session, AST_CAUSE_PROTOCOL_ERROR);
			ast_log(LOG_ERROR, "Incomplete ICE-UDP candidate received on session '%s'\n", session->sid);
			return -1;
		}

		if ((sscanf(component, "%30u", &local_candidate.id) != 1) ||
		    (sscanf(priority, "%30u", (unsigned *)&local_candidate.priority) != 1) ||
		    (sscanf(port, "%30d", &real_port) != 1)) {
			jingle_queue_hangup_with_cause(session, AST_CAUSE_PROTOCOL_ERROR);
			ast_log(LOG_ERROR, "Invalid ICE-UDP candidate information received on session '%s'\n", session->sid);
			return -1;
		}

		local_candidate.foundation = foundation;
		local_candidate.transport = protocol;

		ast_sockaddr_parse(&local_candidate.address, ip, PARSE_PORT_FORBID);

		/* We only support IPv4 right now */
		if (!ast_sockaddr_is_ipv4(&local_candidate.address)) {
			continue;
		}

		ast_sockaddr_set_port(&local_candidate.address, real_port);

		if (!strcasecmp(type, "host")) {
			local_candidate.type = AST_RTP_ICE_CANDIDATE_TYPE_HOST;
		} else if (!strcasecmp(type, "srflx")) {
			local_candidate.type = AST_RTP_ICE_CANDIDATE_TYPE_SRFLX;
		} else if (!strcasecmp(type, "relay")) {
			local_candidate.type = AST_RTP_ICE_CANDIDATE_TYPE_RELAYED;
		} else {
			continue;
		}

		/* Worst case use the first viable address */
		ast_rtp_instance_get_remote_address(rtp, &remote_address);

		if (ast_sockaddr_is_ipv4(&local_candidate.address) && ast_sockaddr_isnull(&remote_address)) {
			ast_rtp_instance_set_remote_address(rtp, &local_candidate.address);
		}

		ice->add_remote_candidate(rtp, &local_candidate);
	}

	ice->start(rtp);

	return 0;
}

/*! \brief Helper function which handles Google transport information */
static int jingle_interpret_google_transport(struct jingle_session *session, iks *transport, struct ast_rtp_instance *rtp)
{
	struct ast_rtp_engine_ice *ice = ast_rtp_instance_get_ice(rtp);
	iks *candidate;

	if (!ice) {
		jingle_queue_hangup_with_cause(session, AST_CAUSE_SWITCH_CONGESTION);
		ast_log(LOG_ERROR, "Received Google transport information on session '%s' but ICE support not available\n", session->sid);
		return -1;
	}

	/* If this session has not transitioned to the Google transport do so now */
	if ((session->transport != JINGLE_TRANSPORT_GOOGLE_V2) &&
	    (session->transport != JINGLE_TRANSPORT_GOOGLE_V1)) {
		/* Stop built-in ICE support... we need to fall back to the old old old STUN */
		ice->stop(rtp);

		session->transport = JINGLE_TRANSPORT_GOOGLE_V2;
	}

	for (candidate = iks_child(transport); candidate; candidate = iks_next(candidate)) {
		char *address = iks_find_attrib(candidate, "address"), *port = iks_find_attrib(candidate, "port");
		char *username = iks_find_attrib(candidate, "username"), *name = iks_find_attrib(candidate, "name");
		char *protocol = iks_find_attrib(candidate, "protocol");
		int real_port;
		struct ast_sockaddr target = { { 0, } };
		/* In Google land the combined value is 32 bytes */
		char combined[33] = "";

		/* If this is NOT actually a candidate just skip it */
		if (strcasecmp(iks_name(candidate), "candidate") &&
		    strcasecmp(iks_name(candidate), "p:candidate") &&
		    strcasecmp(iks_name(candidate), "ses:candidate")) {
			continue;
		}

		/* If this candidate is incomplete skip it */
		if (ast_strlen_zero(address) || ast_strlen_zero(port) || ast_strlen_zero(username) ||
		    ast_strlen_zero(name)) {
			jingle_queue_hangup_with_cause(session, AST_CAUSE_PROTOCOL_ERROR);
			ast_log(LOG_ERROR, "Incomplete Google candidate received on session '%s'\n", session->sid);
			return -1;
		}

		/* We only support UDP so skip any other protocols */
		if (!ast_strlen_zero(protocol) && strcasecmp(protocol, "udp")) {
			continue;
		}

		/* We only permit audio and video, not RTCP */
		if (strcasecmp(name, "rtp") && strcasecmp(name, "video_rtp")) {
			continue;
		}

		/* Parse the target information so we can send a STUN request to the candidate */
		if (sscanf(port, "%30d", &real_port) != 1) {
			jingle_queue_hangup_with_cause(session, AST_CAUSE_PROTOCOL_ERROR);
			ast_log(LOG_ERROR, "Invalid Google candidate port '%s' received on session '%s'\n", port, session->sid);
			return -1;
		}
		ast_sockaddr_parse(&target, address, PARSE_PORT_FORBID);
		ast_sockaddr_set_port(&target, real_port);

		/* Per the STUN support Google talk uses combine the two usernames */
		snprintf(combined, sizeof(combined), "%s%s", username, ice->get_ufrag(rtp));

		/* This should appease the masses... we will actually change the remote address when we get their STUN packet */
		ast_rtp_instance_stun_request(rtp, &target, combined);
	}

	return 0;
}

/*!
 * \brief Helper function which locates content stanzas and interprets them
 *
 * \note The session *must not* be locked before calling this
 */
static int jingle_interpret_content(struct jingle_session *session, ikspak *pak)
{
	iks *content;
	unsigned int changed = 0;
	struct ast_channel *chan;

	/* Look at the content in the session initiation */
	for (content = iks_child(iks_child(pak->x)); content; content = iks_next(content)) {
		char *name;
		struct ast_rtp_instance *rtp = NULL;
		iks *description, *transport;

		/* Ignore specific parts if they are known not to be useful */
		if (!strcmp(iks_name(content), "conference-info")) {
			continue;
		}

		name = iks_find_attrib(content, "name");

		if (session->transport != JINGLE_TRANSPORT_GOOGLE_V1) {
			/* If this content stanza has no name consider it invalid and move on */
			if (ast_strlen_zero(name) && !(name = iks_find_attrib(content, "jin:name"))) {
				jingle_queue_hangup_with_cause(session, AST_CAUSE_BEARERCAPABILITY_NOTAVAIL);
				ast_log(LOG_ERROR, "Received content without a name on session '%s'\n", session->sid);
				return -1;
			}

			/* Try to pre-populate which RTP instance this content is relevant to */
			if (!strcmp(session->audio_name, name)) {
				rtp = session->rtp;
			} else if (!strcmp(session->video_name, name)) {
				rtp = session->vrtp;
			}
		} else {
			/* Google-V1 has no concept of assocating things like the above does, so since we only support audio over it assume they want audio */
			rtp = session->rtp;
		}

		/* If description information is available use it */
		if ((description = iks_find_with_attrib(content, "description", "xmlns", JINGLE_RTP_NS)) ||
		    (description = iks_find_with_attrib(content, "rtp:description", "xmlns:rtp", JINGLE_RTP_NS)) ||
		    (description = iks_find_with_attrib(content, "pho:description", "xmlns:pho", GOOGLE_PHONE_NS)) ||
		    (description = iks_find_with_attrib(pak->query, "description", "xmlns", GOOGLE_PHONE_NS)) ||
		    (description = iks_find_with_attrib(pak->query, "pho:description", "xmlns:pho", GOOGLE_PHONE_NS)) ||
		    (description = iks_find_with_attrib(pak->query, "vid:description", "xmlns", GOOGLE_VIDEO_NS))) {
			/* If we failed to do something with the content description abort immediately */
			if (jingle_interpret_description(session, description, name, &rtp)) {
				return -1;
			}

			/* If we successfully interpret the description then the codecs need updating */
			changed = 1;
		}

		/* If we get past the description handling and we still don't know what RTP instance this is for... it is unknown content */
		if (!rtp) {
			ast_log(LOG_ERROR, "Received a content stanza but have no RTP instance for it on session '%s'\n", session->sid);
			jingle_queue_hangup_with_cause(session, AST_CAUSE_SWITCH_CONGESTION);
			return -1;
		}

		/* If ICE UDP transport information is available use it */
		if ((transport = iks_find_with_attrib(content, "transport", "xmlns", JINGLE_ICE_UDP_NS))) {
			if (jingle_interpret_ice_udp_transport(session, transport, rtp)) {
				return -1;
			}
		} else if ((transport = iks_find_with_attrib(content, "transport", "xmlns", GOOGLE_TRANSPORT_NS)) ||
			   (transport = iks_find_with_attrib(content, "p:transport", "xmlns:p", GOOGLE_TRANSPORT_NS)) ||
			   (transport = iks_find_with_attrib(pak->x, "session", "xmlns", GOOGLE_SESSION_NS)) ||
			   (transport = iks_find_with_attrib(pak->x, "ses:session", "xmlns:ses", GOOGLE_SESSION_NS))) {
			/* If Google transport support is available use it */
			if (jingle_interpret_google_transport(session, transport, rtp)) {
				return -1;
			}
		} else if (iks_find(content, "transport")) {
			/* If this is a transport we do not support terminate the session as it probably won't work out in the end */
			jingle_queue_hangup_with_cause(session, AST_CAUSE_FACILITY_NOT_IMPLEMENTED);
			ast_log(LOG_ERROR, "Unsupported transport type received on session '%s'\n", session->sid);
			return -1;
		}
	}

	if (!changed) {
		return 0;
	}

	if ((chan = jingle_session_lock_full(session))) {
		struct ast_format fmt;

		ast_format_cap_copy(ast_channel_nativeformats(chan), session->jointcap);
		ast_codec_choose(&session->prefs, session->jointcap, 1, &fmt);
		ast_set_read_format(chan, &fmt);
		ast_set_write_format(chan, &fmt);

		ast_channel_unlock(chan);
		ast_channel_unref(chan);
	}
	ao2_unlock(session);

	return 0;
}

/*! \brief Handler function for the 'session-initiate' action */
static void jingle_action_session_initiate(struct jingle_endpoint *endpoint, struct jingle_session *session, ikspak *pak)
{
	char *sid;
	enum jingle_transport transport = JINGLE_TRANSPORT_NONE;
	struct ast_channel *chan;
	int res;

	if (session) {
		/* This is a duplicate session setup, so respond accordingly */
		jingle_send_error_response(endpoint->connection, pak, "result", "out-of-order", NULL);
		return;
	}

	/* Retrieve the session identifier from the message, note that this may alter the transport */
	if ((sid = iks_find_attrib(pak->query, "id"))) {
		/* The presence of the session identifier in the 'id' attribute tells us that this is Google-V1 as everything else uses 'sid' */
		transport = JINGLE_TRANSPORT_GOOGLE_V1;
	} else if (!(sid = iks_find_attrib(pak->query, "sid"))) {
		jingle_send_error_response(endpoint->connection, pak, "bad-request", NULL, NULL);
		return;
	}

	/* Create a new local session */
	if (!(session = jingle_alloc(endpoint, pak->from->full, sid))) {
		jingle_send_error_response(endpoint->connection, pak, "cancel", "service-unavailable xmlns='urn:ietf:params:xml:ns:xmpp-stanzas'", NULL);
		return;
	}

	/* If we determined that the transport should change as a result of how we got the SID change it */
	if (transport != JINGLE_TRANSPORT_NONE) {
		session->transport = transport;
	}

	/* Create a new Asterisk channel using the above local session */
	if (!(chan = jingle_new(endpoint, session, AST_STATE_DOWN, pak->from->user, NULL, pak->from->full))) {
		ao2_ref(session, -1);
		jingle_send_error_response(endpoint->connection, pak, "cancel", "service-unavailable xmlns='urn:ietf:params:xml:ns:xmpp-stanzas'", NULL);
		return;
	}

	ao2_link(endpoint->state->sessions, session);

	ast_setstate(chan, AST_STATE_RING);
	res = ast_pbx_start(chan);

	switch (res) {
	case AST_PBX_FAILED:
		ast_log(LOG_WARNING, "Failed to start PBX :(\n");
		jingle_send_error_response(endpoint->connection, pak, "cancel", "service-unavailable xmlns='urn:ietf:params:xml:ns:xmpp-stanzas'", NULL);
		session->gone = 1;
		ast_hangup(chan);
		break;
	case AST_PBX_CALL_LIMIT:
		ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
		jingle_send_error_response(endpoint->connection, pak, "wait", "resource-constraint xmlns='urn:ietf:params:xml:ns:xmpp-stanzas'", NULL);
		ast_hangup(chan);
		break;
	case AST_PBX_SUCCESS:
		jingle_send_response(endpoint->connection, pak);

		/* Only send a transport-info message if we successfully interpreted the available content */
		if (!jingle_interpret_content(session, pak)) {
			jingle_send_transport_info(session, iks_find_attrib(pak->x, "from"));
		}
		break;
	}
}

/*! \brief Handler function for the 'transport-info' action */
static void jingle_action_transport_info(struct jingle_endpoint *endpoint, struct jingle_session *session, ikspak *pak)
{
	if (!session) {
		jingle_send_error_response(endpoint->connection, pak, "cancel", "item-not-found xmlns='urn:ietf:params:xml:ns:xmpp-stanzas'",
					   "unknown-session xmlns='urn:xmpp:jingle:errors:1'");
		return;
	}

	jingle_interpret_content(session, pak);
	jingle_send_response(endpoint->connection, pak);
}

/*! \brief Handler function for the 'session-accept' action */
static void jingle_action_session_accept(struct jingle_endpoint *endpoint, struct jingle_session *session, ikspak *pak)
{
	struct ast_channel *chan;

	if (!session) {
		jingle_send_error_response(endpoint->connection, pak, "cancel", "item-not-found xmlns='urn:ietf:params:xml:ns:xmpp-stanzas'",
					   "unknown-session xmlns='urn:xmpp:jingle:errors:1'");
		return;
	}


	jingle_interpret_content(session, pak);

	if ((chan = jingle_session_lock_full(session))) {
		ast_queue_control(chan, AST_CONTROL_ANSWER);
		ast_channel_unlock(chan);
		ast_channel_unref(chan);
	}
	ao2_unlock(session);

	jingle_send_response(endpoint->connection, pak);
}

/*! \brief Handler function for the 'session-info' action */
static void jingle_action_session_info(struct jingle_endpoint *endpoint, struct jingle_session *session, ikspak *pak)
{
	struct ast_channel *chan;

	if (!session) {
		jingle_send_error_response(endpoint->connection, pak, "cancel", "item-not-found xmlns='urn:ietf:params:xml:ns:xmpp-stanzas'",
					   "unknown-session xmlns='urn:xmpp:jingle:errors:1'");
		return;
	}

	if (!(chan = jingle_session_lock_full(session))) {
		ao2_unlock(session);
		jingle_send_response(endpoint->connection, pak);
		return;
	}

	if (iks_find_with_attrib(pak->query, "ringing", "xmlns", JINGLE_RTP_INFO_NS)) {
		ast_queue_control(chan, AST_CONTROL_RINGING);
		if (ast_channel_state(chan) != AST_STATE_UP) {
			ast_setstate(chan, AST_STATE_RINGING);
		}
	} else if (iks_find_with_attrib(pak->query, "hold", "xmlns", JINGLE_RTP_INFO_NS)) {
		ast_queue_control(chan, AST_CONTROL_HOLD);
	} else if (iks_find_with_attrib(pak->query, "unhold", "xmlns", JINGLE_RTP_INFO_NS)) {
		ast_queue_control(chan, AST_CONTROL_UNHOLD);
	}

	ast_channel_unlock(chan);
	ast_channel_unref(chan);
	ao2_unlock(session);

	jingle_send_response(endpoint->connection, pak);
}

/*! \brief Handler function for the 'session-terminate' action */
static void jingle_action_session_terminate(struct jingle_endpoint *endpoint, struct jingle_session *session, ikspak *pak)
{
	struct ast_channel *chan;
	iks *reason, *text;
	int cause = AST_CAUSE_NORMAL;
	struct ast_control_pvt_cause_code *cause_code;
	int data_size = sizeof(*cause_code);

	if (!session) {
		jingle_send_error_response(endpoint->connection, pak, "cancel", "item-not-found xmlns='urn:ietf:params:xml:ns:xmpp-stanzas'",
					   "unknown-session xmlns='urn:xmpp:jingle:errors:1'");
		return;
	}

	if (!(chan = jingle_session_lock_full(session))) {
		ao2_unlock(session);
		jingle_send_response(endpoint->connection, pak);
		return;
	}

	/* Pull the reason text from the session-terminate message and translate it into a cause code */
	if ((reason = iks_find(pak->query, "reason")) && (text = iks_child(reason))) {
		int i;

		/* Size of the string making up the cause code is "Motif " + text */
		data_size += 6 + strlen(iks_name(text));
		cause_code = ast_alloca(data_size);
		memset(cause_code, 0, data_size);

		/* Get the appropriate cause code mapping for this reason */
		for (i = 0; i < ARRAY_LEN(jingle_reason_mappings); i++) {
			if (!strcasecmp(jingle_reason_mappings[i].reason, iks_name(text))) {
				cause = jingle_reason_mappings[i].cause;
				break;
			}
		}

		/* Store the technology specific information */
		snprintf(cause_code->code, data_size - sizeof(*cause_code) + 1, "Motif %s", iks_name(text));
	} else {
		/* No technology specific information is available */
		cause_code = ast_alloca(data_size);
		memset(cause_code, 0, data_size);
	}

	ast_copy_string(cause_code->chan_name, ast_channel_name(chan), AST_CHANNEL_NAME);
	cause_code->ast_cause = cause;
	ast_queue_control_data(chan, AST_CONTROL_PVT_CAUSE_CODE, cause_code, data_size);
	ast_channel_hangupcause_hash_set(chan, cause_code, data_size);

	ast_debug(3, "Hanging up channel '%s' due to session terminate message with cause '%d'\n", ast_channel_name(chan), cause);
	ast_queue_hangup_with_cause(chan, cause);
	session->gone = 1;

	ast_channel_unlock(chan);
	ast_channel_unref(chan);
	ao2_unlock(session);

	jingle_send_response(endpoint->connection, pak);
}

/*! \brief Callback for when a Jingle action is received from an endpoint */
static int jingle_action_hook(void *data, ikspak *pak)
{
	char *action;
	const char *sid = NULL;
	struct jingle_session *session = NULL;
	struct jingle_endpoint *endpoint = data;
	int i, handled = 0;

	/* We accept both Jingle and Google-V1 */
	if (!(action = iks_find_attrib(pak->query, "action")) &&
	    !(action = iks_find_attrib(pak->query, "type"))) {
		/* This occurs if either receive a packet masquerading as Jingle or Google-V1 that is actually not OR we receive a response
		 * to a message that has no response hook. */
		return IKS_FILTER_EAT;
	}

	/* Bump the endpoint reference count up in case a reload occurs. Unfortunately the available synchronization between iksemel and us
	 * does not permit us to make this completely safe. */
	ao2_ref(endpoint, +1);

	/* If a Jingle session identifier is present use it */
	if (!(sid = iks_find_attrib(pak->query, "sid"))) {
		/* If a Google-V1 session identifier is present use it */
		sid = iks_find_attrib(pak->query, "id");
	}

	/* If a session identifier was present in the message attempt to find the session, it is up to the action handler whether
	 * this is required or not */
	if (!ast_strlen_zero(sid)) {
		session = ao2_find(endpoint->state->sessions, sid, OBJ_KEY);
	}

	/* If a session is present associate the callid with this thread */
	if (session) {
		ast_callid_threadassoc_add(session->callid);
	}

	/* Iterate through supported action handlers looking for one that is able to handle this */
	for (i = 0; i < ARRAY_LEN(jingle_action_handlers); i++) {
		if (!strcasecmp(jingle_action_handlers[i].action, action)) {
			jingle_action_handlers[i].handler(endpoint, session, pak);
			handled = 1;
			break;
		}
	}

	/* If no action handler is present for the action they sent us make it evident */
	if (!handled) {
		ast_log(LOG_NOTICE, "Received action '%s' for session '%s' that has no handler\n", action, sid);
	}

	/* If a session was successfully found for this message deref it now since the handler is done */
	if (session) {
		ast_callid_threadassoc_remove();
		ao2_ref(session, -1);
	}

	ao2_ref(endpoint, -1);

	return IKS_FILTER_EAT;
}

/*! \brief Custom handler for groups */
static int custom_group_handler(const struct aco_option *opt, struct ast_variable *var, void *obj)
{
	struct jingle_endpoint *endpoint = obj;

	if (!strcasecmp(var->name, "callgroup")) {
		endpoint->callgroup = ast_get_group(var->value);
	} else if (!strcasecmp(var->name, "pickupgroup")) {
		endpoint->pickupgroup = ast_get_group(var->value);
	} else {
		return -1;
	}

	return 0;
}

/*! \brief Custom handler for connection */
static int custom_connection_handler(const struct aco_option *opt, struct ast_variable *var, void *obj)
{
	struct jingle_endpoint *endpoint = obj;

	/* You might think... but Josh, shouldn't you do this in a prelink callback? Well I *could* but until the original is destroyed
	 * this will not actually get called, so even if the config turns out to be bogus this is harmless.
	 */
	if (!(endpoint->connection = ast_xmpp_client_find(var->value))) {
		ast_log(LOG_ERROR, "Connection '%s' configured on endpoint '%s' could not be found\n", var->value, endpoint->name);
		return -1;
	}

	if (!(endpoint->rule = iks_filter_add_rule(endpoint->connection->filter, jingle_action_hook, endpoint,
						   IKS_RULE_TYPE, IKS_PAK_IQ,
						   IKS_RULE_NS, JINGLE_NS,
						   IKS_RULE_NS, GOOGLE_SESSION_NS,
						   IKS_RULE_DONE))) {
		ast_log(LOG_ERROR, "Action hook could not be added to connection '%s' on endpoint '%s'\n", var->value, endpoint->name);
		return -1;
	}

	return 0;
}

/*! \brief Custom handler for transport */
static int custom_transport_handler(const struct aco_option *opt, struct ast_variable *var, void *obj)
{
	struct jingle_endpoint *endpoint = obj;

	if (!strcasecmp(var->value, "ice-udp")) {
		endpoint->transport = JINGLE_TRANSPORT_ICE_UDP;
	} else if (!strcasecmp(var->value, "google")) {
		endpoint->transport = JINGLE_TRANSPORT_GOOGLE_V2;
	} else if (!strcasecmp(var->value, "google-v1")) {
		endpoint->transport = JINGLE_TRANSPORT_GOOGLE_V1;
	} else {
		ast_log(LOG_WARNING, "Unknown transport type '%s' on endpoint '%s', defaulting to 'ice-udp'\n", var->value, endpoint->name);
		endpoint->transport = JINGLE_TRANSPORT_ICE_UDP;
	}

	return 0;
}

/*! \brief Load module into PBX, register channel */
static int load_module(void)
{
	if (!(jingle_tech.capabilities = ast_format_cap_alloc())) {
		return AST_MODULE_LOAD_DECLINE;
	}

	if (aco_info_init(&cfg_info)) {
		ast_log(LOG_ERROR, "Unable to intialize configuration for chan_motif.\n");
		goto end;
	}

	aco_option_register(&cfg_info, "context", ACO_EXACT, endpoint_options, "default", OPT_STRINGFIELD_T, 0, STRFLDSET(struct jingle_endpoint, context));
	aco_option_register_custom(&cfg_info, "callgroup", ACO_EXACT, endpoint_options, NULL, custom_group_handler, 0);
	aco_option_register_custom(&cfg_info, "pickupgroup", ACO_EXACT, endpoint_options, NULL, custom_group_handler, 0);
	aco_option_register(&cfg_info, "language", ACO_EXACT, endpoint_options, NULL, OPT_STRINGFIELD_T, 0, STRFLDSET(struct jingle_endpoint, language));
	aco_option_register(&cfg_info, "musicclass", ACO_EXACT, endpoint_options, NULL, OPT_STRINGFIELD_T, 0, STRFLDSET(struct jingle_endpoint, musicclass));
	aco_option_register(&cfg_info, "parkinglot", ACO_EXACT, endpoint_options, NULL, OPT_STRINGFIELD_T, 0, STRFLDSET(struct jingle_endpoint, parkinglot));
	aco_option_register(&cfg_info, "accountcode", ACO_EXACT, endpoint_options, NULL, OPT_STRINGFIELD_T, 0, STRFLDSET(struct jingle_endpoint, accountcode));
	aco_option_register(&cfg_info, "allow", ACO_EXACT, endpoint_options, "ulaw,alaw", OPT_CODEC_T, 1, FLDSET(struct jingle_endpoint, prefs, cap));
	aco_option_register(&cfg_info, "disallow", ACO_EXACT, endpoint_options, "all", OPT_CODEC_T, 0, FLDSET(struct jingle_endpoint, prefs, cap));
	aco_option_register_custom(&cfg_info, "connection", ACO_EXACT, endpoint_options, NULL, custom_connection_handler, 0);
	aco_option_register_custom(&cfg_info, "transport", ACO_EXACT, endpoint_options, NULL, custom_transport_handler, 0);
	aco_option_register(&cfg_info, "maxicecandidates", ACO_EXACT, endpoint_options, DEFAULT_MAX_ICE_CANDIDATES, OPT_UINT_T, PARSE_DEFAULT,
			    FLDSET(struct jingle_endpoint, maxicecandidates), DEFAULT_MAX_ICE_CANDIDATES);
	aco_option_register(&cfg_info, "maxpayloads", ACO_EXACT, endpoint_options, DEFAULT_MAX_PAYLOADS, OPT_UINT_T, PARSE_DEFAULT,
			    FLDSET(struct jingle_endpoint, maxpayloads), DEFAULT_MAX_PAYLOADS);

	ast_format_cap_add_all_by_type(jingle_tech.capabilities, AST_FORMAT_TYPE_AUDIO);

	if (aco_process_config(&cfg_info, 0)) {
		ast_log(LOG_ERROR, "Unable to read config file motif.conf. Not loading module.\n");
		aco_info_destroy(&cfg_info);
		ast_format_cap_destroy(jingle_tech.capabilities);
		jingle_tech.capabilities = NULL;
		return AST_MODULE_LOAD_DECLINE;
	}

	if (!(sched = ast_sched_context_create())) {
		ast_log(LOG_ERROR, "Unable to create scheduler context.\n");
		goto end;
	}

	if (ast_sched_start_thread(sched)) {
		ast_log(LOG_ERROR, "Unable to create scheduler context thread.\n");
		goto end;
	}

	ast_rtp_glue_register(&jingle_rtp_glue);

	if (ast_channel_register(&jingle_tech)) {
		ast_log(LOG_ERROR, "Unable to register channel class %s\n", channel_type);
		goto end;
	}

	return 0;

end:
	ast_rtp_glue_unregister(&jingle_rtp_glue);

	if (sched) {
		ast_sched_context_destroy(sched);
	}

	aco_info_destroy(&cfg_info);
	ao2_global_obj_release(globals);

	ast_format_cap_destroy(jingle_tech.capabilities);
	jingle_tech.capabilities = NULL;

	return AST_MODULE_LOAD_FAILURE;
}

/*! \brief Reload module */
static int reload(void)
{
	return aco_process_config(&cfg_info, 1);
}

/*! \brief Unload the jingle channel from Asterisk */
static int unload_module(void)
{
	ast_channel_unregister(&jingle_tech);
	ast_format_cap_destroy(jingle_tech.capabilities);
	jingle_tech.capabilities = NULL;
	ast_rtp_glue_unregister(&jingle_rtp_glue);
	ast_sched_context_destroy(sched);
	aco_info_destroy(&cfg_info);
	ao2_global_obj_release(globals);

	return 0;
}

AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Motif Jingle Channel Driver",
		.load = load_module,
		.unload = unload_module,
		.reload = reload,
		.load_pri = AST_MODPRI_CHANNEL_DRIVER,
	       );