/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2025, Sangoma Technologies Corporation
 *
 * George Joseph <gjoseph@sangoma.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 George Joseph <gjoseph@sangoma.com>
 *
 * \brief Websocket Media Channel
 *
 * \ingroup channel_drivers
 */

/*** MODULEINFO
	<depend>res_http_websocket</depend>
	<depend>res_websocket_client</depend>
	<support_level>core</support_level>
 ***/

#include "asterisk.h"

#include "asterisk/app.h"
#include "asterisk/causes.h"
#include "asterisk/channel.h"
#include "asterisk/codec.h"
#include "asterisk/http_websocket.h"
#include "asterisk/format_cache.h"
#include "asterisk/frame.h"
#include "asterisk/lock.h"
#include "asterisk/mod_format.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/uuid.h"
#include "asterisk/timing.h"
#include "asterisk/translate.h"
#include "asterisk/websocket_client.h"

static struct ast_websocket_server *ast_ws_server;

static struct ao2_container *instances = NULL;

struct websocket_pvt {
	enum ast_websocket_type type;
	struct ast_websocket_client *client;
	struct ast_websocket *websocket;
	struct ast_format *native_format;
	struct ast_codec *native_codec;
	struct ast_format *slin_format;
	struct ast_codec *slin_codec;
	struct ast_channel *channel;
	struct ast_timer *timer;
	struct ast_frame silence;
	struct ast_trans_pvt *translator;
	AST_LIST_HEAD(, ast_frame) frame_queue;
	pthread_t outbound_read_thread;
	size_t bytes_read;
	size_t leftover_len;
	char *leftover_data;
	int no_auto_answer;
	int optimal_frame_size;
	int bulk_media_in_progress;
	int report_queue_drained;
	int frame_queue_length;
	int queue_full;
	int queue_paused;
	char connection_id[0];
};

#define MEDIA_WEBSOCKET_OPTIMAL_FRAME_SIZE "MEDIA_WEBSOCKET_OPTIMAL_FRAME_SIZE"
#define MEDIA_WEBSOCKET_CONNECTION_ID "MEDIA_WEBSOCKET_CONNECTION_ID"
#define INCOMING_CONNECTION_ID "INCOMING"

#define ANSWER_CHANNEL "ANSWER"
#define HANGUP_CHANNEL "HANGUP"
#define START_MEDIA_BUFFERING "START_MEDIA_BUFFERING"
#define STOP_MEDIA_BUFFERING "STOP_MEDIA_BUFFERING"
#define FLUSH_MEDIA "FLUSH_MEDIA"
#define GET_DRIVER_STATUS "GET_STATUS"
#define REPORT_QUEUE_DRAINED "REPORT_QUEUE_DRAINED"
#define PAUSE_MEDIA "PAUSE_MEDIA"
#define CONTINUE_MEDIA "CONTINUE_MEDIA"

#define MEDIA_START "MEDIA_START"
#define MEDIA_XON "MEDIA_XON"
#define MEDIA_XOFF "MEDIA_XOFF"
#define QUEUE_DRAINED "QUEUE_DRAINED"
#define DRIVER_STATUS "STATUS"
#define MEDIA_BUFFERING_COMPLETED "MEDIA_BUFFERING_COMPLETED"

#define QUEUE_LENGTH_MAX 1000
#define QUEUE_LENGTH_XOFF_LEVEL 900
#define QUEUE_LENGTH_XON_LEVEL 800
#define MAX_TEXT_MESSAGE_LEN MIN(128, (AST_WEBSOCKET_MAX_RX_PAYLOAD_SIZE - 1))

/* Forward declarations */
static struct ast_channel *webchan_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause);
static int webchan_call(struct ast_channel *ast, const char *dest, int timeout);
static struct ast_frame *webchan_read(struct ast_channel *ast);
static int webchan_write(struct ast_channel *ast, struct ast_frame *f);
static int webchan_hangup(struct ast_channel *ast);

static struct ast_channel_tech websocket_tech = {
	.type = "WebSocket",
	.description = "Media over WebSocket Channel Driver",
	.requester = webchan_request,
	.call = webchan_call,
	.read = webchan_read,
	.write = webchan_write,
	.hangup = webchan_hangup,
};

static void set_channel_format(struct websocket_pvt * instance,
	struct ast_format *fmt)
{
	if (ast_format_cmp(ast_channel_rawreadformat(instance->channel), fmt)
		== AST_FORMAT_CMP_NOT_EQUAL) {
		ast_channel_set_rawreadformat(instance->channel, fmt);
		ast_debug(4, "Switching readformat to %s\n", ast_format_get_name(fmt));
	}
}

/*
 * Reminder...  This function gets called by webchan_read which is
 * triggered by the channel timer firing.  It always gets called
 * every 20ms (or whatever the timer is set to) even if there are
 * no frames in the queue.
 */
static struct ast_frame *dequeue_frame(struct websocket_pvt *instance)
{
	struct ast_frame *queued_frame = NULL;
	SCOPED_LOCK(frame_queue_lock, &instance->frame_queue, AST_LIST_LOCK,
		AST_LIST_UNLOCK);

	/*
	 * If the queue is paused, don't read a frame.  Processing
	 * will continue down the function and a silence frame will
	 * be sent in its place.
	 */
	if (instance->queue_paused) {
		return NULL;
	}

	/*
	 * We need to check if we need to send an XON before anything
	 * else because there are multiple escape paths in this function
	 * and we don't want to accidentally keep the queue in a "full"
	 * state.
	 */
	if (instance->queue_full && instance->frame_queue_length < QUEUE_LENGTH_XON_LEVEL) {
		instance->queue_full = 0;
		ast_debug(4, "%s: WebSocket sending MEDIA_XON\n",
			ast_channel_name(instance->channel));
		ast_websocket_write_string(instance->websocket, MEDIA_XON);
	}

	queued_frame = AST_LIST_REMOVE_HEAD(&instance->frame_queue, frame_list);

	/*
	 * If there are no frames in the queue, we need to
	 * return NULL so we can send a silence frame.  We also need
	 * to send the QUEUE_DRAINED notification if we were requested
	 * to do so.
	 */
	if (!queued_frame) {
		if (instance->report_queue_drained) {
			instance->report_queue_drained = 0;
			ast_debug(4, "%s: WebSocket sending QUEUE_DRAINED\n",
				ast_channel_name(instance->channel));
			ast_websocket_write_string(instance->websocket, QUEUE_DRAINED);
		}
		return NULL;
	}

	/*
	 * The only way a control frame could be present here is as
	 * a result of us calling queue_option_frame() in response
	 * to an incoming TEXT command from the websocket.
	 * We'll be safe and make sure it's a AST_CONTROL_OPTION
	 * frame anyway.
	 *
	 * It's quite possible that there are multiple control frames
	 * in a row in the queue so we need to process consecutive ones
	 * immediately.
	 *
	 * In any case, processing a control frame MUST not use up
	 * a media timeslot so after all control frames have been
	 * processed, we need to read an audio frame and process it.
	 */
	while (queued_frame && queued_frame->frametype == AST_FRAME_CONTROL) {
		if (queued_frame->subclass.integer == AST_CONTROL_OPTION) {
			/*
			 * We just need to send the data to the websocket.
			 * The data should already be NULL terminated.
			 */
			ast_websocket_write_string(instance->websocket,
				queued_frame->data.ptr);
			ast_debug(4, "%s: WebSocket sending %s\n",
				ast_channel_name(instance->channel), (char *)queued_frame->data.ptr);
		}
		/*
		 * We do NOT send these to the core so we need to free
		 * the frame and grab the next one.  If it's also a
		 * control frame, we need to process it otherwise
		 * continue down in the function.
		 */
		ast_frame_free(queued_frame, 0);
		queued_frame = AST_LIST_REMOVE_HEAD(&instance->frame_queue, frame_list);
		/*
		 * Jut FYI... We didn't bump the queue length when we added the control
		 * frames so we don't need to decrement it here.
		 */
	}

	/*
	 * If, after reading all control frames,  there are no frames
	 * left in the queue, we need to return NULL so we can send
	 * a silence frame.
	 */
	if (!queued_frame) {
		return NULL;
	}

	instance->frame_queue_length--;

	return queued_frame;
}
/*!
 * \internal
 *
 * Called by the core channel thread each time the instance timer fires.
 *
 */
static struct ast_frame *webchan_read(struct ast_channel *ast)
{
	struct websocket_pvt *instance = NULL;
	struct ast_frame *native_frame = NULL;
	struct ast_frame *slin_frame = NULL;

	instance = ast_channel_tech_pvt(ast);
	if (!instance) {
		return NULL;
	}

	if (ast_timer_get_event(instance->timer) == AST_TIMING_EVENT_EXPIRED) {
		ast_timer_ack(instance->timer, 1);
	}

	native_frame = dequeue_frame(instance);

	/*
	 * No frame when the timer fires means we have to create and
	 * return a silence frame in its place.
	 */
	if (!native_frame) {
		ast_debug(5, "%s: WebSocket read timer fired with no frame available.  Returning silence.\n", ast_channel_name(ast));
		set_channel_format(instance, instance->slin_format);
		slin_frame = ast_frdup(&instance->silence);
		return slin_frame;
	}

	/*
	 * If the frame length is already optimal_frame_size, we can just
	 * return it.
	 */
	if (native_frame->datalen == instance->optimal_frame_size) {
		set_channel_format(instance, instance->native_format);
		return native_frame;
	}

	/*
	 * If we're here, we have a short frame that we need to pad
	 * with silence.
	 */

	if (instance->translator) {
		slin_frame = ast_translate(instance->translator, native_frame, 0);
		if (!slin_frame) {
			ast_log(LOG_WARNING, "%s: Failed to translate %d byte frame\n",
				ast_channel_name(ast), native_frame->datalen);
			return NULL;
		}
		ast_frame_free(native_frame, 0);
	} else {
		/*
		 * If there was no translator then the native format
		 * was already slin.
		 */
		slin_frame = native_frame;
	}

	set_channel_format(instance, instance->slin_format);

	/*
	 * So now we have an slin frame but it's probably still short
	 * so we create a new data buffer with the correct length
	 * which is filled with zeros courtesy of ast_calloc.
	 * We then copy the short frame data into the new buffer
	 * and set the offset to AST_FRIENDLY_OFFSET so that
	 * the core can read the data without any issues.
	 * If the original frame data was mallocd, we need to free the old
	 * data buffer so we don't leak memory and we need to set
	 * mallocd to AST_MALLOCD_DATA so that the core knows
	 * it needs to free the new data buffer when it's done.
	 */

	if (slin_frame->datalen != instance->silence.datalen) {
		char *old_data = slin_frame->data.ptr;
		int old_len = slin_frame->datalen;
		int old_offset = slin_frame->offset;
		ast_debug(4, "%s: WebSocket read short frame. Expected %d got %d.  Filling with silence\n",
			ast_channel_name(ast), instance->silence.datalen,
			slin_frame->datalen);

		slin_frame->data.ptr = ast_calloc(1, instance->silence.datalen + AST_FRIENDLY_OFFSET);
		if (!slin_frame->data.ptr) {
			ast_frame_free(slin_frame, 0);
			return NULL;
		}
		slin_frame->data.ptr += AST_FRIENDLY_OFFSET;
		slin_frame->offset = AST_FRIENDLY_OFFSET;
		memcpy(slin_frame->data.ptr, old_data, old_len);
		if (slin_frame->mallocd & AST_MALLOCD_DATA) {
			ast_free(old_data - old_offset);
		}
		slin_frame->mallocd |= AST_MALLOCD_DATA;
		slin_frame->datalen = instance->silence.datalen;
		slin_frame->samples = instance->silence.samples;
	}

	return slin_frame;
}

static int queue_frame_from_buffer(struct websocket_pvt *instance,
	char *buffer, size_t len)
{
	struct ast_frame fr = { 0, };
	struct ast_frame *duped_frame = NULL;

	AST_FRAME_SET_BUFFER(&fr, buffer, 0, len);
	fr.frametype = AST_FRAME_VOICE;
	fr.subclass.format = instance->native_format;
	fr.samples = instance->native_codec->samples_count(&fr);

	duped_frame = ast_frisolate(&fr);
	if (!duped_frame) {
		ast_log(LOG_WARNING, "%s: Failed to isolate frame\n",
			ast_channel_name(instance->channel));
		return -1;
	}

	{
		SCOPED_LOCK(frame_queue_lock, &instance->frame_queue, AST_LIST_LOCK,
			AST_LIST_UNLOCK);
		AST_LIST_INSERT_TAIL(&instance->frame_queue, duped_frame, frame_list);
		instance->frame_queue_length++;
		if (!instance->queue_full && instance->frame_queue_length >= QUEUE_LENGTH_XOFF_LEVEL) {
			instance->queue_full = 1;
			ast_debug(4, "%s: WebSocket sending %s\n",
				ast_channel_name(instance->channel), MEDIA_XOFF);
			ast_websocket_write_string(instance->websocket, MEDIA_XOFF);
		}
	}

	ast_debug(5, "%s: Queued %d byte frame\n", ast_channel_name(instance->channel),
		duped_frame->datalen);

	return 0;
}

static int queue_option_frame(struct websocket_pvt *instance,
	char *buffer)
{
	struct ast_frame fr = { 0, };
	struct ast_frame *duped_frame = NULL;

	AST_FRAME_SET_BUFFER(&fr, buffer, 0, strlen(buffer) + 1);
	fr.frametype = AST_FRAME_CONTROL;
	fr.subclass.integer = AST_CONTROL_OPTION;

	duped_frame = ast_frisolate(&fr);
	if (!duped_frame) {
		ast_log(LOG_WARNING, "%s: Failed to isolate frame\n",
			ast_channel_name(instance->channel));
		return -1;
	}

	AST_LIST_LOCK(&instance->frame_queue);
	AST_LIST_INSERT_TAIL(&instance->frame_queue, duped_frame, frame_list);
	AST_LIST_UNLOCK(&instance->frame_queue);

	ast_debug(4, "%s: Queued '%s' option frame\n",
		ast_channel_name(instance->channel), buffer);

	return 0;
}

static int process_text_message(struct websocket_pvt *instance,
	char *payload, uint64_t payload_len)
{
	int res = 0;
	char *command;

	if (payload_len > MAX_TEXT_MESSAGE_LEN) {
		ast_log(LOG_WARNING, "%s: WebSocket TEXT message of length %d exceeds maximum length of %d\n",
			ast_channel_name(instance->channel), (int)payload_len, MAX_TEXT_MESSAGE_LEN);
		return 0;
	}

	/*
	 * This is safe because the payload buffer is always >= 8K
	 * even with LOW_MEMORY defined and we've already made sure the
	 * command is less than 128 bytes.
	 */
	payload[payload_len] = '\0';
	command = ast_strip(ast_strdupa(payload));

	ast_debug(4, "%s: WebSocket %s command received\n",
		ast_channel_name(instance->channel), command);

	if (ast_strings_equal(command, ANSWER_CHANNEL)) {
		ast_queue_control(instance->channel, AST_CONTROL_ANSWER);

	} else if (ast_strings_equal(command, HANGUP_CHANNEL)) {
		ast_queue_control(instance->channel, AST_CONTROL_HANGUP);

	} else if (ast_strings_equal(command, START_MEDIA_BUFFERING)) {
		AST_LIST_LOCK(&instance->frame_queue);
		instance->bulk_media_in_progress = 1;
		AST_LIST_UNLOCK(&instance->frame_queue);

	} else if (ast_begins_with(command, STOP_MEDIA_BUFFERING)) {
		char *id;
		char *option;
		SCOPED_LOCK(frame_queue_lock, &instance->frame_queue, AST_LIST_LOCK,
			AST_LIST_UNLOCK);

		id = ast_strip(command + strlen(STOP_MEDIA_BUFFERING));

		ast_debug(4, "%s: WebSocket %s '%s' with %d bytes in leftover_data.\n",
			ast_channel_name(instance->channel), STOP_MEDIA_BUFFERING, id,
			(int)instance->leftover_len);

		instance->bulk_media_in_progress = 0;
		if (instance->leftover_len > 0) {
			res = queue_frame_from_buffer(instance, instance->leftover_data, instance->leftover_len);
			if (res != 0) {
				return res;
			}
		}
		instance->leftover_len = 0;
		res = ast_asprintf(&option, "%s%s%s", MEDIA_BUFFERING_COMPLETED,
			S_COR(!ast_strlen_zero(id), " ", ""), S_OR(id, ""));
		if (res <= 0 || !option) {
			return res;
		}
		res = queue_option_frame(instance, option);
		ast_free(option);

	} else if (ast_strings_equal(command, FLUSH_MEDIA)) {
		struct ast_frame *frame = NULL;
		AST_LIST_LOCK(&instance->frame_queue);
		while ((frame = AST_LIST_REMOVE_HEAD(&instance->frame_queue, frame_list))) {
			ast_frfree(frame);
		}
		instance->frame_queue_length = 0;
		instance->bulk_media_in_progress = 0;
		instance->leftover_len = 0;
		AST_LIST_UNLOCK(&instance->frame_queue);

	} else if (ast_strings_equal(payload, REPORT_QUEUE_DRAINED)) {
		AST_LIST_LOCK(&instance->frame_queue);
		instance->report_queue_drained = 1;
		AST_LIST_UNLOCK(&instance->frame_queue);

	} else if (ast_strings_equal(command, GET_DRIVER_STATUS)) {
		char *status = NULL;

		res = ast_asprintf(&status, "%s queue_length:%d xon_level:%d xoff_level:%d queue_full:%s bulk_media:%s media_paused:%s",
			DRIVER_STATUS,
			instance->frame_queue_length, QUEUE_LENGTH_XON_LEVEL,
			QUEUE_LENGTH_XOFF_LEVEL,
			S_COR(instance->queue_full, "true", "false"),
			S_COR(instance->bulk_media_in_progress, "true", "false"),
			S_COR(instance->queue_paused, "true", "false")
			);
		if (res <= 0 || !status) {
			ast_free(status);
			res = -1;
		} else {
			ast_debug(4, "%s: WebSocket status: %s\n",
				ast_channel_name(instance->channel), status);
			res = ast_websocket_write_string(instance->websocket, status);
			ast_free(status);
		}

	} else if (ast_strings_equal(payload, PAUSE_MEDIA)) {
		AST_LIST_LOCK(&instance->frame_queue);
		instance->queue_paused = 1;
		AST_LIST_UNLOCK(&instance->frame_queue);

	} else if (ast_strings_equal(payload, CONTINUE_MEDIA)) {
		AST_LIST_LOCK(&instance->frame_queue);
		instance->queue_paused = 0;
		AST_LIST_UNLOCK(&instance->frame_queue);

	} else {
		ast_log(LOG_WARNING, "%s: WebSocket %s command unknown\n",
			ast_channel_name(instance->channel), command);
	}

	return res;
}

static int process_binary_message(struct websocket_pvt *instance,
	char *payload, uint64_t payload_len)
{
	char *next_frame_ptr = NULL;
	size_t bytes_read = 0;
	int res = 0;
	size_t bytes_left = 0;

	{
		SCOPED_LOCK(frame_queue_lock, &instance->frame_queue, AST_LIST_LOCK,
			AST_LIST_UNLOCK);
		if (instance->frame_queue_length >= QUEUE_LENGTH_MAX) {
			ast_debug(4, "%s: WebSocket queue is full. Ignoring incoming binary message.\n",
				ast_channel_name(instance->channel));
			return 0;
		}
	}

	next_frame_ptr = payload;
	instance->bytes_read += payload_len;

	if (instance->bulk_media_in_progress && instance->leftover_len > 0) {
		/*
		 * We have leftover data from a previous websocket message.
		 * Try to make a complete frame by appending data from
		 * the current message to the leftover data.
		 */
		char *append_ptr = instance->leftover_data + instance->leftover_len;
		size_t bytes_needed_for_frame = instance->optimal_frame_size - instance->leftover_len;
		/*
		 * It's possible that even the current message doesn't have enough
		 * data to make a complete frame.
		 */
		size_t bytes_avail_to_copy = MIN(bytes_needed_for_frame, payload_len);

		/*
		 * Append whatever we can to the end of the leftover data
		 * even if it's not enough to make a complete frame.
		 */
		memcpy(append_ptr, payload, bytes_avail_to_copy);

		/*
		 * If leftover data is still short, just return and wait for the
		 * next websocket message.
		 */
		if (bytes_avail_to_copy < bytes_needed_for_frame) {
			ast_debug(4, "%s: Leftover data %d bytes but only %d new bytes available of %d needed. Appending and waiting for next message.\n",
				ast_channel_name(instance->channel), (int)instance->leftover_len, (int)bytes_avail_to_copy, (int)bytes_needed_for_frame);
			instance->leftover_len += bytes_avail_to_copy;
			return 0;
		}

		res = queue_frame_from_buffer(instance, instance->leftover_data, instance->optimal_frame_size);
		if (res < 0) {
			return -1;
		}

		/*
		 * We stole data from the current payload so decrement payload_len
		 * and set the next frame pointer after the data in payload
		 * we just copied.
		 */
		payload_len -= bytes_avail_to_copy;
		next_frame_ptr = payload + bytes_avail_to_copy;

		ast_debug(5, "%s: --- BR: %4d  FQ: %4d  PL: %4d  LOL: %3d  P: %p  NFP: %p  OFF: %4d  NPL: %4d  BAC: %3d\n",
			ast_channel_name(instance->channel),
			instance->frame_queue_length,
			(int)instance->bytes_read,
			(int)(payload_len + bytes_avail_to_copy),
			(int)instance->leftover_len,
			payload,
			next_frame_ptr,
			(int)(next_frame_ptr - payload),
			(int)payload_len,
			(int)bytes_avail_to_copy
			);


		instance->leftover_len = 0;
	}

	if (!instance->bulk_media_in_progress && instance->leftover_len > 0) {
		instance->leftover_len = 0;
	}

	bytes_left = payload_len;
	while (bytes_read < payload_len && bytes_left >= instance->optimal_frame_size) {
		res = queue_frame_from_buffer(instance, next_frame_ptr,
			instance->optimal_frame_size);
		if (res < 0) {
			break;
		}
		bytes_read += instance->optimal_frame_size;
		next_frame_ptr += instance->optimal_frame_size;
		bytes_left -= instance->optimal_frame_size;
	}

	if (instance->bulk_media_in_progress && bytes_left > 0) {
		/*
		 * We have a partial frame.  Save the leftover data.
		 */
		ast_debug(5, "%s: +++ BR: %4d  FQ: %4d  PL: %4d  LOL: %3d  P: %p  NFP: %p  OFF: %4d  BL: %4d\n",
			ast_channel_name(instance->channel),
			(int)instance->bytes_read,
			instance->frame_queue_length,
			(int)payload_len,
			(int)instance->leftover_len,
			payload,
			next_frame_ptr,
			(int)(next_frame_ptr - payload),
			(int)bytes_left
			);
		memcpy(instance->leftover_data, next_frame_ptr, bytes_left);
		instance->leftover_len = bytes_left;
	}

	return 0;
}

static int read_from_ws_and_queue(struct websocket_pvt *instance)
{
	uint64_t payload_len = 0;
	char *payload = NULL;
	enum ast_websocket_opcode opcode;
	int fragmented = 0;
	int res = 0;

	if (!instance || !instance->websocket) {
		ast_log(LOG_WARNING, "%s: WebSocket instance not found\n",
			ast_channel_name(instance->channel));
		return -1;
	}

	ast_debug(9, "%s: Waiting for websocket to have data\n", ast_channel_name(instance->channel));
	res = ast_wait_for_input(
		ast_websocket_fd(instance->websocket), -1);
	if (res <= 0) {
		ast_log(LOG_WARNING, "%s: WebSocket read failed: %s\n",
			ast_channel_name(instance->channel), strerror(errno));
		return -1;
	}

	/*
	 * We need to lock here to prevent the websocket handle from
	 * being pulled out from under us if the core sends us a
	 * hangup request.
	 */
	ao2_lock(instance);
	if (!instance->websocket) {
		ao2_unlock(instance);
		return -1;
	}

	res = ast_websocket_read(instance->websocket, &payload, &payload_len,
		&opcode, &fragmented);
	ao2_unlock(instance);
	if (res) {
		return -1;
	}
	ast_debug(5, "%s: WebSocket read %d bytes\n", ast_channel_name(instance->channel),
		(int)payload_len);

	if (opcode == AST_WEBSOCKET_OPCODE_TEXT) {
		return process_text_message(instance, payload, payload_len);
	}

	if (opcode == AST_WEBSOCKET_OPCODE_CLOSE) {
		ast_debug(5, "%s: WebSocket closed by remote\n",
			ast_channel_name(instance->channel));
		return -1;
	}

	if (opcode != AST_WEBSOCKET_OPCODE_BINARY) {
		ast_debug(5, "%s: WebSocket frame type %d not supported. Ignoring.\n",
			ast_channel_name(instance->channel), (int)opcode);
		return 0;
	}

	return process_binary_message(instance, payload, payload_len);
}

/*!
 * \internal
 *
 * For incoming websocket connections, this function gets called by
 * incoming_ws_established_cb() and is run in the http server thread
 * handling the websocket connection.
 *
 * For outgoing websocket connections, this function gets started as
 * a background thread by webchan_call().
 */
static void *read_thread_handler(void *obj)
{
	RAII_VAR(struct websocket_pvt *, instance, obj, ao2_cleanup);
	RAII_VAR(char *, command, NULL, ast_free);
	int res = 0;

	ast_debug(3, "%s: Read thread started\n", ast_channel_name(instance->channel));

	/*
	 * We need to tell the remote app what channel this media is for.
	 * This is especially important for outbound connections otherwise
	 * the app won't know who the media is for.
	 */
	res = ast_asprintf(&command, "%s connection_id:%s channel:%s format:%s optimal_frame_size:%d", MEDIA_START,
		instance->connection_id, ast_channel_name(instance->channel),
		ast_format_get_name(instance->native_format),
		instance->optimal_frame_size);
	if (res <= 0 || !command) {
		ast_queue_control(instance->channel, AST_CONTROL_HANGUP);
		ast_log(LOG_ERROR, "%s: Failed to create MEDIA_START\n", ast_channel_name(instance->channel));
		return NULL;
	}
	res = ast_websocket_write_string(instance->websocket, command);
	if (res != 0) {
		ast_log(LOG_ERROR, "%s: Failed to send MEDIA_START\n", ast_channel_name(instance->channel));
		ast_queue_control(instance->channel, AST_CONTROL_HANGUP);
		return NULL;
	}
	ast_debug(3, "%s: Sent %s\n", ast_channel_name(instance->channel),
		command);

	if (!instance->no_auto_answer) {
		ast_debug(3, "%s: ANSWER by auto_answer\n", ast_channel_name(instance->channel));
		ast_queue_control(instance->channel, AST_CONTROL_ANSWER);
	}

	while (read_from_ws_and_queue(instance) == 0)
	{
	}

	/*
	 * websocket_hangup will take care of closing the websocket if needed.
	 */
	ast_debug(3, "%s: HANGUP by websocket close/error\n", ast_channel_name(instance->channel));
	ast_queue_control(instance->channel, AST_CONTROL_HANGUP);

	return NULL;
}

/*! \brief Function called when we should write a frame to the channel */
static int webchan_write(struct ast_channel *ast, struct ast_frame *f)
{
	struct websocket_pvt *instance = ast_channel_tech_pvt(ast);

	if (!instance || !instance->websocket) {
		ast_log(LOG_WARNING, "%s: WebSocket instance or client not found\n",
			ast_channel_name(ast));
		return -1;
	}

	if (f->frametype != AST_FRAME_VOICE) {
		ast_log(LOG_WARNING, "%s: This WebSocket channel only supports AST_FRAME_VOICE frames\n",
			ast_channel_name(ast));
		return -1;
	}
	if (f->subclass.format != instance->native_format) {
		ast_log(LOG_WARNING, "%s: This WebSocket channel only supports the '%s' format\n",
			ast_channel_name(ast), ast_format_get_name(instance->native_format));
		return -1;
	}

	return ast_websocket_write(instance->websocket, AST_WEBSOCKET_OPCODE_BINARY,
		(char *)f->data.ptr, (uint64_t)f->datalen);
}

/*!
 * \internal
 *
 * Called by the core to actually call the remote.
 */
static int webchan_call(struct ast_channel *ast, const char *dest,
	int timeout)
{
	struct websocket_pvt *instance = ast_channel_tech_pvt(ast);
	int nodelay = 1;
	enum ast_websocket_result result;

	if (!instance) {
		ast_log(LOG_WARNING, "%s: WebSocket instance not found\n",
			ast_channel_name(ast));
		return -1;
	}

	if (instance->type == AST_WS_TYPE_SERVER) {
		ast_debug(3, "%s: Websocket call incoming\n", ast_channel_name(instance->channel));
		return 0;
	}
	ast_debug(3, "%s: Websocket call outgoing\n", ast_channel_name(instance->channel));

	if (!instance->client) {
		ast_log(LOG_WARNING, "%s: WebSocket client not found\n",
			ast_channel_name(ast));
		return -1;
	}

	ast_debug(3, "%s: WebSocket call requested to %s. cid: %s\n",
		ast_channel_name(ast), dest, instance->connection_id);

	instance->websocket = ast_websocket_client_connect(instance->client,
		instance, ast_channel_name(ast), &result);
	if (!instance->websocket || result != WS_OK) {
		ast_log(LOG_WARNING, "%s: WebSocket connection failed to %s: %s\n",
			ast_channel_name(ast), dest, ast_websocket_result_to_str(result));
		return -1;
	}

	if (setsockopt(ast_websocket_fd(instance->websocket),
		IPPROTO_TCP, TCP_NODELAY, (char *) &nodelay, sizeof(nodelay)) < 0) {
		ast_log(LOG_WARNING, "Failed to set TCP_NODELAY on websocket connection: %s\n", strerror(errno));
	}

	ast_debug(3, "%s: WebSocket connection to %s established\n",
		ast_channel_name(ast), dest);

	/* read_thread_handler() will clean up the bump */
	if (ast_pthread_create_detached_background(&instance->outbound_read_thread, NULL,
			read_thread_handler, ao2_bump(instance))) {
		ast_log(LOG_WARNING, "%s: Failed to create thread.\n", ast_channel_name(ast));
		ao2_cleanup(instance);
		return -1;
	}

	return 0;
}

static void websocket_destructor(void *data)
{
	struct websocket_pvt *instance = data;
	struct ast_frame *frame = NULL;
	ast_debug(3, "%s: WebSocket instance freed\n", instance->connection_id);

	AST_LIST_LOCK(&instance->frame_queue);
	while ((frame = AST_LIST_REMOVE_HEAD(&instance->frame_queue, frame_list))) {
		ast_frfree(frame);
	}
	AST_LIST_UNLOCK(&instance->frame_queue);

	if (instance->timer) {
		ast_timer_close(instance->timer);
		instance->timer = NULL;
	}

	if (instance->channel) {
		ast_channel_unref(instance->channel);
		instance->channel = NULL;
	}
	if (instance->websocket) {
		ast_websocket_unref(instance->websocket);
		instance->websocket = NULL;
	}

	ao2_cleanup(instance->client);
	instance->client = NULL;

	ao2_cleanup(instance->native_codec);
	instance->native_codec = NULL;

	ao2_cleanup(instance->native_format);
	instance->native_format = NULL;

	ao2_cleanup(instance->slin_codec);
	instance->slin_codec = NULL;

	ao2_cleanup(instance->slin_format);
	instance->slin_format = NULL;

	if (instance->silence.data.ptr) {
		ast_free(instance->silence.data.ptr);
		instance->silence.data.ptr = NULL;
	}

	if (instance->translator) {
		ast_translator_free_path(instance->translator);
		instance->translator = NULL;
	}

	if (instance->leftover_data) {
		ast_free(instance->leftover_data);
		instance->leftover_data = NULL;
	}
}

struct instance_proxy {
	AO2_WEAKPROXY();
	/*! \brief The name of the module owning this sorcery instance */
	char connection_id[0];
};

static void instance_proxy_cb(void *weakproxy, void *data)
{
	struct instance_proxy *proxy = weakproxy;
	ast_debug(3, "%s: WebSocket instance removed from instances\n", proxy->connection_id);
	ao2_unlink(instances, weakproxy);
}

static struct websocket_pvt* websocket_new(const char *chan_name,
	const char *connection_id, struct ast_format *fmt)
{
	RAII_VAR(struct instance_proxy *, proxy, NULL, ao2_cleanup);
	RAII_VAR(struct websocket_pvt *, instance, NULL, ao2_cleanup);
	char uuid[AST_UUID_STR_LEN];
	enum ast_websocket_type ws_type;

	SCOPED_AO2WRLOCK(locker, instances);

	if (ast_strings_equal(connection_id, INCOMING_CONNECTION_ID)) {
		connection_id = ast_uuid_generate_str(uuid, sizeof(uuid));
		ws_type = AST_WS_TYPE_SERVER;
	} else {
		ws_type = AST_WS_TYPE_CLIENT;
	}

	proxy = ao2_weakproxy_alloc(sizeof(*proxy) + strlen(connection_id) + 1, NULL);
	if (!proxy) {
		return NULL;
	}
	strcpy(proxy->connection_id, connection_id); /* Safe */

	instance = ao2_alloc(sizeof(*instance) + strlen(connection_id) + 1,
		websocket_destructor);
	if (!instance) {
		return NULL;
	}
	strcpy(instance->connection_id, connection_id); /* Safe */

	instance->type = ws_type;
	if (ws_type == AST_WS_TYPE_CLIENT) {
		instance->client = ast_websocket_client_retrieve_by_id(instance->connection_id);
		if (!instance->client) {
			ast_log(LOG_ERROR, "%s: WebSocket client connection '%s' not found\n",
				chan_name, instance->connection_id);
			return NULL;
		}
	}

	AST_LIST_HEAD_INIT(&instance->frame_queue);

	/*
	 * We need the codec to calculate the number of samples in a frame
	 * so we'll get it once and store it in the instance.
	 *
	 * References for native_format and native_codec are now held by the
	 * instance and will be released when the instance is destroyed.
	 */
	instance->native_format = fmt;
	instance->native_codec = ast_format_get_codec(instance->native_format);
	/*
	 * References for native_format and native_codec are now held by the
	 * instance and will be released when the instance is destroyed.
	 */
	instance->optimal_frame_size =
		(instance->native_codec->default_ms * instance->native_codec->minimum_bytes)
			/ instance->native_codec->minimum_ms;

	instance->leftover_data = ast_calloc(1, instance->optimal_frame_size);
	if (!instance->leftover_data) {
		return NULL;
	}

	/* We have exclusive access to proxy and sorcery, no need for locking here. */
	if (ao2_weakproxy_set_object(proxy, instance, OBJ_NOLOCK)) {
		return NULL;
	}

	if (ao2_weakproxy_subscribe(proxy, instance_proxy_cb, NULL, OBJ_NOLOCK)) {
		return NULL;
	}

	if (!ao2_link_flags(instances, proxy, OBJ_NOLOCK)) {
		ast_log(LOG_ERROR, "%s: Unable to link WebSocket instance to instances\n",
			proxy->connection_id);
		return NULL;
	}
	ast_debug(3, "%s: WebSocket instance created and linked\n", proxy->connection_id);

	return ao2_bump(instance);
}

static int set_instance_translator(struct websocket_pvt *instance)
{
	if (ast_format_cache_is_slinear(instance->native_format)) {
		instance->slin_format = ao2_bump(instance->native_format);
		instance->slin_codec = ast_format_get_codec(instance->slin_format);
		return 0;
	}

	instance->slin_format = ao2_bump(ast_format_cache_get_slin_by_rate(instance->native_codec->sample_rate));
	if (!instance->slin_format) {
		ast_log(LOG_ERROR, "%s: Unable to get slin format for rate %d\n",
			ast_channel_name(instance->channel), instance->native_codec->sample_rate);
		return -1;
	}
	ast_debug(3, "%s: WebSocket channel slin format '%s' Sample rate: %d ptime: %dms\n",
		ast_channel_name(instance->channel), ast_format_get_name(instance->slin_format),
		ast_format_get_sample_rate(instance->slin_format),
		ast_format_get_default_ms(instance->slin_format));

	instance->translator = ast_translator_build_path(instance->slin_format, instance->native_format);
	if (!instance->translator) {
		ast_log(LOG_ERROR, "%s: Unable to build translator path from '%s' to '%s'\n",
			ast_channel_name(instance->channel), ast_format_get_name(instance->native_format),
			ast_format_get_name(instance->slin_format));
		return -1;
	}

	instance->slin_codec = ast_format_get_codec(instance->slin_format);
	return 0;
}

static int set_instance_silence_frame(struct websocket_pvt *instance)
{
	instance->silence.frametype = AST_FRAME_VOICE;
	instance->silence.datalen =
		(instance->slin_codec->default_ms * instance->slin_codec->minimum_bytes) / instance->slin_codec->minimum_ms;
	instance->silence.samples = instance->silence.datalen / sizeof(uint16_t);
	/*
	 * Even though we'll calloc the data pointer, we don't mark it as
	 * mallocd because this frame will be around for a while and we don't
	 * want it accidentally freed before we're done with it.
	 */
	instance->silence.mallocd = 0;
	instance->silence.offset = 0;
	instance->silence.src = __PRETTY_FUNCTION__;
	instance->silence.subclass.format = instance->slin_format;
	instance->silence.data.ptr = ast_calloc(1, instance->silence.datalen);
	if (!instance->silence.data.ptr) {
		return -1;
	}

	return 0;
}

static int set_channel_timer(struct websocket_pvt *instance)
{
	int rate = 0;
	instance->timer = ast_timer_open();
	if (!instance->timer) {
		return -1;
	}
	/* Rate is the number of ticks per second, not the interval. */
	rate = 1000 / ast_format_get_default_ms(instance->native_format);
	ast_debug(3, "%s: WebSocket timer rate %d\n",
		ast_channel_name(instance->channel), rate);
	ast_timer_set_rate(instance->timer, rate);
	/*
	 * Calling ast_channel_set_fd will cause the channel thread to call
	 * webchan_read at 'rate' times per second.
	 */
	ast_channel_set_fd(instance->channel, 0, ast_timer_fd(instance->timer));

	return 0;
}

static int set_channel_variables(struct websocket_pvt *instance)
{
	char *pkt_size = NULL;
	int res = ast_asprintf(&pkt_size, "%d", instance->optimal_frame_size);
	if (res <= 0) {
		return -1;
	}

	pbx_builtin_setvar_helper(instance->channel, MEDIA_WEBSOCKET_OPTIMAL_FRAME_SIZE,
		pkt_size);
	ast_free(pkt_size);
	pbx_builtin_setvar_helper(instance->channel, MEDIA_WEBSOCKET_CONNECTION_ID,
		instance->connection_id);

	return 0;
}

enum {
	OPT_WS_CODEC =  (1 << 0),
	OPT_WS_NO_AUTO_ANSWER =  (1 << 1),
};

enum {
	OPT_ARG_WS_CODEC,
	OPT_ARG_WS_NO_AUTO_ANSWER,
	OPT_ARG_ARRAY_SIZE
};

AST_APP_OPTIONS(websocket_options, BEGIN_OPTIONS
	AST_APP_OPTION_ARG('c', OPT_WS_CODEC, OPT_ARG_WS_CODEC),
	AST_APP_OPTION('n', OPT_WS_NO_AUTO_ANSWER),
	END_OPTIONS );

static struct ast_channel *webchan_request(const char *type,
	struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids,
	const struct ast_channel *requestor, const char *data, int *cause)
{
	char *parse;
	RAII_VAR(struct websocket_pvt *, instance, NULL, ao2_cleanup);
	struct ast_channel *chan = NULL;
	struct ast_format *fmt = NULL;
	struct ast_format_cap *caps = NULL;
	AST_DECLARE_APP_ARGS(args,
		AST_APP_ARG(connection_id);
		AST_APP_ARG(options);
	);
	struct ast_flags opts = { 0, };
	char *opt_args[OPT_ARG_ARRAY_SIZE];
	const char *requestor_name = requestor ? ast_channel_name(requestor) : "no channel";

	ast_debug(3, "%s: WebSocket channel requested\n",
		requestor_name);

	if (ast_strlen_zero(data)) {
		ast_log(LOG_ERROR, "%s: A connection id is required for the 'WebSocket' channel\n",
			requestor_name);
		goto failure;
	}
	parse = ast_strdupa(data);
	AST_NONSTANDARD_APP_ARGS(args, parse, '/');

	if (ast_strlen_zero(args.connection_id)) {
		ast_log(LOG_ERROR, "%s: connection_id is required for the 'WebSocket' channel\n",
			requestor_name);
		goto failure;
	}

	if (!ast_strlen_zero(args.options)
		&& ast_app_parse_options(websocket_options, &opts, opt_args,
			ast_strdupa(args.options))) {
		ast_log(LOG_ERROR, "%s: 'WebSocket' channel options '%s' parse error\n",
			requestor_name, args.options);
		goto failure;
	}

	if (ast_test_flag(&opts, OPT_WS_CODEC)
		&& !ast_strlen_zero(opt_args[OPT_ARG_WS_CODEC])) {
		ast_debug(3, "%s: Using specified format %s\n",
			requestor_name, opt_args[OPT_ARG_WS_CODEC]);
		fmt = ast_format_cache_get(opt_args[OPT_ARG_WS_CODEC]);
	} else {
		/*
		 * If codec wasn't specified in the dial string,
		 * use the first format in the capabilities.
		 */
		ast_debug(3, "%s: Using format %s from requesting channel\n",
			requestor_name, opt_args[OPT_ARG_WS_CODEC]);
		fmt = ast_format_cap_get_format(cap, 0);
	}

	if (!fmt) {
		ast_log(LOG_WARNING, "%s: No codec found for sending media to connection '%s'\n",
			requestor_name, args.connection_id);
		goto failure;
	}

	instance = websocket_new(requestor_name, args.connection_id, fmt);
	if (!instance) {
		ast_log(LOG_ERROR, "%s: Failed to allocate WebSocket channel pvt\n",
			requestor_name);
		goto failure;
	}

	instance->no_auto_answer = ast_test_flag(&opts, OPT_WS_NO_AUTO_ANSWER);

	chan = ast_channel_alloc(1, AST_STATE_DOWN, "", "", "", "", "", assignedids,
		requestor, 0, "WebSocket/%s/%p", args.connection_id, instance);
	if (!chan) {
		ast_log(LOG_ERROR, "%s: Unable to alloc channel\n", ast_channel_name(requestor));
		goto failure;
	}

	ast_debug(3, "%s: WebSocket channel %s allocated for connection %s\n",
		ast_channel_name(chan), requestor_name,
		instance->connection_id);

	instance->channel = ao2_bump(chan);
	ast_channel_tech_set(instance->channel, &websocket_tech);

	if (set_instance_translator(instance) != 0) {
		goto failure;
	}

	if (set_instance_silence_frame(instance) != 0) {
		goto failure;
	}

	if (set_channel_timer(instance) != 0) {
		goto failure;
	}

	if (set_channel_variables(instance) != 0) {
		goto failure;
	}

	caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
	if (!caps) {
		ast_log(LOG_ERROR, "%s: Unable to alloc caps\n", requestor_name);
		goto failure;
	}

	ast_format_cap_append(caps, instance->native_format, 0);
	ast_channel_nativeformats_set(instance->channel, caps);
	ast_channel_set_writeformat(instance->channel, instance->native_format);
	ast_channel_set_rawwriteformat(instance->channel, instance->native_format);
	ast_channel_set_readformat(instance->channel, instance->native_format);
	ast_channel_set_rawreadformat(instance->channel, instance->native_format);
	ast_channel_tech_pvt_set(chan, ao2_bump(instance));
	ast_channel_unlock(chan);
	ao2_cleanup(caps);

	ast_debug(3, "%s: WebSocket channel created to %s\n",
		ast_channel_name(chan), args.connection_id);

	return chan;

failure:
	if (chan) {
		ast_channel_unlock(chan);
	}
	*cause = AST_CAUSE_FAILURE;
	return NULL;
}


/*!
 * \internal
 *
 * Called by the core to hang up the channel.
 */
static int webchan_hangup(struct ast_channel *ast)
{
	struct websocket_pvt *instance = ast_channel_tech_pvt(ast);

	if (!instance) {
		return -1;
	}
	ast_debug(3, "%s: WebSocket call hangup. cid: %s\n",
		ast_channel_name(ast), instance->connection_id);

	/*
	 * We need to lock because read_from_ws_and_queue() is probably waiting
	 * on the websocket file descriptor and will unblock and immediately try to
	 * check the websocket and read from it. We don't want to pull the
	 * websocket out from under it between the check and read.
	 */
	ao2_lock(instance);
	if (instance->websocket) {
		ast_websocket_close(instance->websocket, 1000);
		ast_websocket_unref(instance->websocket);
		instance->websocket = NULL;
	}
	ast_channel_tech_pvt_set(ast, NULL);
	ao2_unlock(instance);

	/* Clean up the reference from adding the instance to the channel */
	ao2_cleanup(instance);

	return 0;
}

/*!
 * \internal
 *
 * Called by res_http_websocket after a client has connected and
 * successfully upgraded from HTTP to WebSocket.
 *
 * Depends on incoming_ws_http_callback parsing the connection_id from
 * the HTTP request and storing it in get_params.
 */
static void incoming_ws_established_cb(struct ast_websocket *ast_ws_session,
	struct ast_variable *get_params, struct ast_variable *upgrade_headers)
{
	RAII_VAR(struct ast_websocket *, s, ast_ws_session, ast_websocket_unref);
	struct ast_variable *v;
	const char *connection_id = NULL;
	struct websocket_pvt *instance = NULL;
	int nodelay = 1;

	ast_debug(3, "WebSocket established\n");

	for (v = upgrade_headers; v; v = v->next) {
		ast_debug(4, "Header-> %s: %s\n", v->name, v->value);
	}
	for (v = get_params; v; v = v->next) {
		ast_debug(4, " Param-> %s: %s\n", v->name, v->value);
	}

	connection_id = ast_variable_find_in_list(get_params, "CONNECTION_ID");
	if (!connection_id) {
		/*
		 * This can't really happen because websocket_http_callback won't
		 * let it get this far if it can't add the connection_id to the
		 * get_params.
		 * Just in case though...
		 */
		ast_log(LOG_WARNING, "WebSocket connection id not found\n");
		ast_queue_control(instance->channel, AST_CONTROL_HANGUP);
		ast_websocket_close(ast_ws_session, 1000);
		return;
	}

	instance = ao2_weakproxy_find(instances, connection_id, OBJ_SEARCH_KEY | OBJ_NOLOCK, "");
	if (!instance) {
		/*
		 * This also can't really happen because websocket_http_callback won't
		 * let it get this far if it can't find the instance.
		 * Just in case though...
		 */
		ast_log(LOG_WARNING, "%s: WebSocket instance not found\n", connection_id);
		ast_queue_control(instance->channel, AST_CONTROL_HANGUP);
		ast_websocket_close(ast_ws_session, 1000);
		return;
	}
	instance->websocket = ao2_bump(ast_ws_session);

	if (setsockopt(ast_websocket_fd(instance->websocket),
		IPPROTO_TCP, TCP_NODELAY, (char *) &nodelay, sizeof(nodelay)) < 0) {
		ast_log(LOG_WARNING, "Failed to set TCP_NODELAY on manager connection: %s\n", strerror(errno));
	}

	/* read_thread_handler cleans up the bump */
	read_thread_handler(ao2_bump(instance));

	ao2_cleanup(instance);
	ast_debug(3, "WebSocket closed\n");
}

/*!
 * \internal
 *
 * Called by the core http server after a client connects but before
 * the upgrade from HTTP to Websocket.  We need to save the URI in
 * the CONNECTION_ID in a get_param because it contains the connection UUID
 * we gave to the client when they used externalMedia to create the channel.
 * incoming_ws_established_cb() will use this to retrieve the chan_websocket
 * instance.
 */
static int incoming_ws_http_callback(struct ast_tcptls_session_instance *ser,
	const struct ast_http_uri *urih, const char *uri,
	enum ast_http_method method, struct ast_variable *get_params,
	struct ast_variable *headers)
{
	struct ast_http_uri fake_urih = {
		.data = ast_ws_server,
	};
	int res = 0;
	/*
	 * Normally the http server will destroy the get_params
	 * when the session ends but if there weren't any initially
	 * and we create some and add them to the list, the http server
	 * won't know about it so we have to destroy it ourselves.
	 */
	int destroy_get_params = (get_params == NULL);
	struct ast_variable *v = NULL;
	RAII_VAR(struct websocket_pvt *, instance, NULL, ao2_cleanup);

	ast_debug(2, "URI: %s Starting\n", uri);

	/*
	 * The client will have issued the GET request with a URI of
	 * /media/<connection_id>
	 *
	 * Since this callback is registered for the /media URI prefix the
	 * http server will strip that off the front of the URI passing in
	 * only the path components after that in the 'uri' parameter.
	 * This should leave only the connection id without a leading '/'.
	 */
	instance = ao2_weakproxy_find(instances, uri, OBJ_SEARCH_KEY | OBJ_NOLOCK, "");
	if (!instance) {
		ast_log(LOG_WARNING, "%s: WebSocket instance not found\n", uri);
		ast_http_error(ser, 404, "Not found", "WebSocket instance not found");
		return -1;
	}

	/*
	 * We don't allow additional connections using the same connection id.
	 */
	if (instance->websocket) {
		ast_log(LOG_WARNING, "%s: Websocket already connected for channel '%s'\n",
			uri, instance->channel ? ast_channel_name(instance->channel) : "unknown");
		ast_http_error(ser, 409, "Conflict", "Another websocket connection exists for this connection id");
		return -1;
	}

	v = ast_variable_new("CONNECTION_ID", uri, "");
	if (!v) {
		ast_http_error(ser, 500, "Server error", "");
		return -1;
	}
	ast_variable_list_append(&get_params, v);

	for (v = get_params; v; v = v->next) {
		ast_debug(4, " Param-> %s: %s\n", v->name, v->value);
	}

	/*
	 * This will ultimately call internal_ws_established_cb() so
	 * this function will block until the websocket is closed and
	 * internal_ws_established_cb() returns;
	 */
	res = ast_websocket_uri_cb(ser, &fake_urih, uri, method,
		get_params, headers);
	if (destroy_get_params) {
		ast_variables_destroy(get_params);
	}

	ast_debug(2, "URI: %s DONE\n", uri);

	return res;
}

static struct ast_http_uri http_uri = {
	.callback = incoming_ws_http_callback,
	.description = "Media over Websocket",
	.uri = "media",
	.has_subtree = 1,
	.data = NULL,
	.key = __FILE__,
	.no_decode_uri = 1,
};

/*! \brief Function called when our module is unloaded */
static int unload_module(void)
{
	ast_http_uri_unlink(&http_uri);
	ao2_cleanup(ast_ws_server);
	ast_ws_server = NULL;

	ast_channel_unregister(&websocket_tech);
	ao2_cleanup(websocket_tech.capabilities);
	websocket_tech.capabilities = NULL;

	ao2_cleanup(instances);
	instances = NULL;

	return 0;
}

AO2_STRING_FIELD_HASH_FN(instance_proxy, connection_id)
AO2_STRING_FIELD_CMP_FN(instance_proxy, connection_id)
AO2_STRING_FIELD_SORT_FN(instance_proxy, connection_id)

/*! \brief Function called when our module is loaded */
static int load_module(void)
{
	int res = 0;
	struct ast_websocket_protocol *protocol;

	if (!(websocket_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
		return AST_MODULE_LOAD_DECLINE;
	}

	ast_format_cap_append_by_type(websocket_tech.capabilities, AST_MEDIA_TYPE_UNKNOWN);
	if (ast_channel_register(&websocket_tech)) {
		ast_log(LOG_ERROR, "Unable to register channel class 'WebSocket'\n");
		unload_module();
		return AST_MODULE_LOAD_DECLINE;
	}

	instances = ao2_container_alloc_hash(AO2_ALLOC_OPT_LOCK_RWLOCK,
		AO2_CONTAINER_ALLOC_OPT_DUPS_REPLACE, 17, instance_proxy_hash_fn,
		instance_proxy_sort_fn, instance_proxy_cmp_fn);
	if (!instances) {
		ast_log(LOG_WARNING,
			"Failed to allocate the chan_websocket instance registry\n");
		unload_module();
		return AST_MODULE_LOAD_DECLINE;
	}

	ast_ws_server = ast_websocket_server_create();
	if (!ast_ws_server) {
		unload_module();
		return AST_MODULE_LOAD_DECLINE;
	}

	protocol = ast_websocket_sub_protocol_alloc("media");
	if (!protocol) {
		unload_module();
		return AST_MODULE_LOAD_DECLINE;
	}
	protocol->session_established = incoming_ws_established_cb;
	res = ast_websocket_server_add_protocol2(ast_ws_server, protocol);

	ast_http_uri_link(&http_uri);

	return res == 0 ? AST_MODULE_LOAD_SUCCESS : AST_MODULE_LOAD_DECLINE;
}

AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Websocket Media Channel",
	.support_level = AST_MODULE_SUPPORT_CORE,
	.load = load_module,
	.unload = unload_module,
	.load_pri = AST_MODPRI_CHANNEL_DRIVER,
	.requires = "res_http_websocket,res_websocket_client",
);