Merge in the RTP engine API.

This API provides a generic way for multiple RTP stacks to be
integrated into Asterisk. Right now there is only one present, res_rtp_asterisk,
which is the existing Asterisk RTP stack. Functionality wise this commit
performs the same as previously. API documentation can be viewed in the
rtp_engine.h header file.

Review: http://reviewboard.digium.com/r/209/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 1572 insertions, 0 deletions

diff --git a/main/rtp_engine.c b/main/rtp_engine.c
new file mode 100644
index 000000000..fd448b849
--- /dev/null
+++ b/main/rtp_engine.c
@@ -0,0 +1,1572 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2008, 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
+ *
+ * \brief Pluggable RTP Architecture
+ *
+ * \author Joshua Colp <jcolp@digium.com>
+ */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include <math.h>
+
+#include "asterisk/channel.h"
+#include "asterisk/frame.h"
+#include "asterisk/module.h"
+#include "asterisk/rtp_engine.h"
+#include "asterisk/manager.h"
+#include "asterisk/options.h"
+#include "asterisk/astobj2.h"
+#include "asterisk/pbx.h"
+
+/*! Structure that represents an RTP session (instance) */
+struct ast_rtp_instance {
+	/*! Engine that is handling this RTP instance */
+	struct ast_rtp_engine *engine;
+	/*! Data unique to the RTP engine */
+	void *data;
+	/*! RTP properties that have been set and their value */
+	int properties[AST_RTP_PROPERTY_MAX];
+	/*! Address that we are expecting RTP to come in to */
+	struct sockaddr_in local_address;
+	/*! Address that we are sending RTP to */
+	struct sockaddr_in remote_address;
+	/*! Instance that we are bridged to if doing remote or local bridging */
+	struct ast_rtp_instance *bridged;
+	/*! Payload and packetization information */
+	struct ast_rtp_codecs codecs;
+	/*! RTP timeout time (negative or zero means disabled, negative value means temporarily disabled) */
+	int timeout;
+	/*! RTP timeout when on hold (negative or zero means disabled, negative value means temporarily disabled). */
+	int holdtimeout;
+	/*! DTMF mode in use */
+	enum ast_rtp_dtmf_mode dtmf_mode;
+};
+
+/*! List of RTP engines that are currently registered */
+static AST_RWLIST_HEAD_STATIC(engines, ast_rtp_engine);
+
+/*! List of RTP glues */
+static AST_RWLIST_HEAD_STATIC(glues, ast_rtp_glue);
+
+/*! The following array defines the MIME Media type (and subtype) for each
+   of our codecs, or RTP-specific data type. */
+static const struct ast_rtp_mime_type {
+	struct ast_rtp_payload_type payload_type;
+	char *type;
+	char *subtype;
+	unsigned int sample_rate;
+} ast_rtp_mime_types[] = {
+	{{1, AST_FORMAT_G723_1}, "audio", "G723", 8000},
+	{{1, AST_FORMAT_GSM}, "audio", "GSM", 8000},
+	{{1, AST_FORMAT_ULAW}, "audio", "PCMU", 8000},
+	{{1, AST_FORMAT_ULAW}, "audio", "G711U", 8000},
+	{{1, AST_FORMAT_ALAW}, "audio", "PCMA", 8000},
+	{{1, AST_FORMAT_ALAW}, "audio", "G711A", 8000},
+	{{1, AST_FORMAT_G726}, "audio", "G726-32", 8000},
+	{{1, AST_FORMAT_ADPCM}, "audio", "DVI4", 8000},
+	{{1, AST_FORMAT_SLINEAR}, "audio", "L16", 8000},
+	{{1, AST_FORMAT_LPC10}, "audio", "LPC", 8000},
+	{{1, AST_FORMAT_G729A}, "audio", "G729", 8000},
+	{{1, AST_FORMAT_G729A}, "audio", "G729A", 8000},
+	{{1, AST_FORMAT_G729A}, "audio", "G.729", 8000},
+	{{1, AST_FORMAT_SPEEX}, "audio", "speex", 8000},
+	{{1, AST_FORMAT_ILBC}, "audio", "iLBC", 8000},
+	/* this is the sample rate listed in the RTP profile for the G.722
+	              codec, *NOT* the actual sample rate of the media stream
+	*/
+	{{1, AST_FORMAT_G722}, "audio", "G722", 8000},
+	{{1, AST_FORMAT_G726_AAL2}, "audio", "AAL2-G726-32", 8000},
+	{{0, AST_RTP_DTMF}, "audio", "telephone-event", 8000},
+	{{0, AST_RTP_CISCO_DTMF}, "audio", "cisco-telephone-event", 8000},
+	{{0, AST_RTP_CN}, "audio", "CN", 8000},
+	{{1, AST_FORMAT_JPEG}, "video", "JPEG", 90000},
+	{{1, AST_FORMAT_PNG}, "video", "PNG", 90000},
+	{{1, AST_FORMAT_H261}, "video", "H261", 90000},
+	{{1, AST_FORMAT_H263}, "video", "H263", 90000},
+	{{1, AST_FORMAT_H263_PLUS}, "video", "h263-1998", 90000},
+	{{1, AST_FORMAT_H264}, "video", "H264", 90000},
+	{{1, AST_FORMAT_MP4_VIDEO}, "video", "MP4V-ES", 90000},
+	{{1, AST_FORMAT_T140RED}, "text", "RED", 1000},
+	{{1, AST_FORMAT_T140}, "text", "T140", 1000},
+	{{1, AST_FORMAT_SIREN7}, "audio", "G7221", 16000},
+	{{1, AST_FORMAT_SIREN14}, "audio", "G7221", 32000},
+};
+
+/*!
+ * \brief Mapping between Asterisk codecs and rtp payload types
+ *
+ * Static (i.e., well-known) RTP payload types for our "AST_FORMAT..."s:
+ * also, our own choices for dynamic payload types.  This is our master
+ * table for transmission
+ *
+ * See http://www.iana.org/assignments/rtp-parameters for a list of
+ * assigned values
+ */
+static const struct ast_rtp_payload_type static_RTP_PT[AST_RTP_MAX_PT] = {
+	[0] = {1, AST_FORMAT_ULAW},
+	#ifdef USE_DEPRECATED_G726
+	[2] = {1, AST_FORMAT_G726}, /* Technically this is G.721, but if Cisco can do it, so can we... */
+	#endif
+	[3] = {1, AST_FORMAT_GSM},
+	[4] = {1, AST_FORMAT_G723_1},
+	[5] = {1, AST_FORMAT_ADPCM}, /* 8 kHz */
+	[6] = {1, AST_FORMAT_ADPCM}, /* 16 kHz */
+	[7] = {1, AST_FORMAT_LPC10},
+	[8] = {1, AST_FORMAT_ALAW},
+	[9] = {1, AST_FORMAT_G722},
+	[10] = {1, AST_FORMAT_SLINEAR}, /* 2 channels */
+	[11] = {1, AST_FORMAT_SLINEAR}, /* 1 channel */
+	[13] = {0, AST_RTP_CN},
+	[16] = {1, AST_FORMAT_ADPCM}, /* 11.025 kHz */
+	[17] = {1, AST_FORMAT_ADPCM}, /* 22.050 kHz */
+	[18] = {1, AST_FORMAT_G729A},
+	[19] = {0, AST_RTP_CN},         /* Also used for CN */
+	[26] = {1, AST_FORMAT_JPEG},
+	[31] = {1, AST_FORMAT_H261},
+	[34] = {1, AST_FORMAT_H263},
+	[97] = {1, AST_FORMAT_ILBC},
+	[98] = {1, AST_FORMAT_H263_PLUS},
+	[99] = {1, AST_FORMAT_H264},
+	[101] = {0, AST_RTP_DTMF},
+	[102] = {1, AST_FORMAT_SIREN7},
+	[103] = {1, AST_FORMAT_H263_PLUS},
+	[104] = {1, AST_FORMAT_MP4_VIDEO},
+	[105] = {1, AST_FORMAT_T140RED},        /* Real time text chat (with redundancy encoding) */
+	[106] = {1, AST_FORMAT_T140},   /* Real time text chat */
+	[110] = {1, AST_FORMAT_SPEEX},
+	[111] = {1, AST_FORMAT_G726},
+	[112] = {1, AST_FORMAT_G726_AAL2},
+	[115] = {1, AST_FORMAT_SIREN14},
+	[121] = {0, AST_RTP_CISCO_DTMF}, /* Must be type 121 */
+};
+
+int ast_rtp_engine_register2(struct ast_rtp_engine *engine, struct ast_module *module)
+{
+	struct ast_rtp_engine *current_engine;
+
+	/* Perform a sanity check on the engine structure to make sure it has the basics */
+	if (ast_strlen_zero(engine->name) || !engine->new || !engine->destroy || !engine->write || !engine->read) {
+		ast_log(LOG_WARNING, "RTP Engine '%s' failed sanity check so it was not registered.\n", !ast_strlen_zero(engine->name) ? engine->name : "Unknown");
+		return -1;
+	}
+
+	/* Link owner module to the RTP engine for reference counting purposes */
+	engine->mod = module;
+
+	AST_RWLIST_WRLOCK(&engines);
+
+	/* Ensure that no two modules with the same name are registered at the same time */
+	AST_RWLIST_TRAVERSE(&engines, current_engine, entry) {
+		if (!strcmp(current_engine->name, engine->name)) {
+			ast_log(LOG_WARNING, "An RTP engine with the name '%s' has already been registered.\n", engine->name);
+			AST_RWLIST_UNLOCK(&engines);
+			return -1;
+		}
+	}
+
+	/* The engine survived our critique. Off to the list it goes to be used */
+	AST_RWLIST_INSERT_TAIL(&engines, engine, entry);
+
+	AST_RWLIST_UNLOCK(&engines);
+
+	ast_verb(2, "Registered RTP engine '%s'\n", engine->name);
+
+	return 0;
+}
+
+int ast_rtp_engine_unregister(struct ast_rtp_engine *engine)
+{
+	struct ast_rtp_engine *current_engine = NULL;
+
+	AST_RWLIST_WRLOCK(&engines);
+
+	if ((current_engine = AST_RWLIST_REMOVE(&engines, engine, entry))) {
+		ast_verb(2, "Unregistered RTP engine '%s'\n", engine->name);
+	}
+
+	AST_RWLIST_UNLOCK(&engines);
+
+	return current_engine ? 0 : -1;
+}
+
+int ast_rtp_glue_register2(struct ast_rtp_glue *glue, struct ast_module *module)
+{
+	struct ast_rtp_glue *current_glue = NULL;
+
+	if (ast_strlen_zero(glue->type)) {
+		return -1;
+	}
+
+	glue->mod = module;
+
+	AST_RWLIST_WRLOCK(&glues);
+
+	AST_RWLIST_TRAVERSE(&glues, current_glue, entry) {
+		if (!strcasecmp(current_glue->type, glue->type)) {
+			ast_log(LOG_WARNING, "RTP glue with the name '%s' has already been registered.\n", glue->type);
+			AST_RWLIST_UNLOCK(&glues);
+			return -1;
+		}
+	}
+
+	AST_RWLIST_INSERT_TAIL(&glues, glue, entry);
+
+	AST_RWLIST_UNLOCK(&glues);
+
+	ast_verb(2, "Registered RTP glue '%s'\n", glue->type);
+
+	return 0;
+}
+
+int ast_rtp_glue_unregister(struct ast_rtp_glue *glue)
+{
+	struct ast_rtp_glue *current_glue = NULL;
+
+	AST_RWLIST_WRLOCK(&glues);
+
+	if ((current_glue = AST_RWLIST_REMOVE(&glues, glue, entry))) {
+		ast_verb(2, "Unregistered RTP glue '%s'\n", glue->type);
+	}
+
+	AST_RWLIST_UNLOCK(&glues);
+
+	return current_glue ? 0 : -1;
+}
+
+static void instance_destructor(void *obj)
+{
+	struct ast_rtp_instance *instance = obj;
+
+	/* Pass us off to the engine to destroy */
+	if (instance->data && instance->engine->destroy(instance)) {
+		ast_debug(1, "Engine '%s' failed to destroy RTP instance '%p'\n", instance->engine->name, instance);
+		return;
+	}
+
+	/* Drop our engine reference */
+	ast_module_unref(instance->engine->mod);
+
+	ast_debug(1, "Destroyed RTP instance '%p'\n", instance);
+}
+
+int ast_rtp_instance_destroy(struct ast_rtp_instance *instance)
+{
+	ao2_ref(instance, -1);
+
+	return 0;
+}
+
+struct ast_rtp_instance *ast_rtp_instance_new(const char *engine_name, struct sched_context *sched, struct sockaddr_in *sin, void *data)
+{
+	struct ast_rtp_instance *instance = NULL;
+	struct ast_rtp_engine *engine = NULL;
+
+	AST_RWLIST_RDLOCK(&engines);
+
+	/* If an engine name was specified try to use it or otherwise use the first one registered */
+	if (!ast_strlen_zero(engine_name)) {
+		AST_RWLIST_TRAVERSE(&engines, engine, entry) {
+			if (!strcmp(engine->name, engine_name)) {
+				break;
+			}
+		}
+	} else {
+		engine = AST_RWLIST_FIRST(&engines);
+	}
+
+	/* If no engine was actually found bail out now */
+	if (!engine) {
+		ast_log(LOG_ERROR, "No RTP engine was found. Do you have one loaded?\n");
+		AST_RWLIST_UNLOCK(&engines);
+		return NULL;
+	}
+
+	/* Bump up the reference count before we return so the module can not be unloaded */
+	ast_module_ref(engine->mod);
+
+	AST_RWLIST_UNLOCK(&engines);
+
+	/* Allocate a new RTP instance */
+	if (!(instance = ao2_alloc(sizeof(*instance), instance_destructor))) {
+		ast_module_unref(engine->mod);
+		return NULL;
+	}
+	instance->engine = engine;
+	memcpy(&instance->local_address, sin, sizeof(instance->local_address));
+
+	ast_debug(1, "Using engine '%s' for RTP instance '%p'\n", engine->name, instance);
+
+	/* And pass it off to the engine to setup */
+	if (instance->engine->new(instance, sched, sin, data)) {
+		ast_debug(1, "Engine '%s' failed to setup RTP instance '%p'\n", engine->name, instance);
+		ao2_ref(instance, -1);
+		return NULL;
+	}
+
+	ast_debug(1, "RTP instance '%p' is setup and ready to go\n", instance);
+
+	return instance;
+}
+
+void ast_rtp_instance_set_data(struct ast_rtp_instance *instance, void *data)
+{
+	instance->data = data;
+}
+
+void *ast_rtp_instance_get_data(struct ast_rtp_instance *instance)
+{
+	return instance->data;
+}
+
+int ast_rtp_instance_write(struct ast_rtp_instance *instance, struct ast_frame *frame)
+{
+	return instance->engine->write(instance, frame);
+}
+
+struct ast_frame *ast_rtp_instance_read(struct ast_rtp_instance *instance, int rtcp)
+{
+	return instance->engine->read(instance, rtcp);
+}
+
+int ast_rtp_instance_set_local_address(struct ast_rtp_instance *instance, struct sockaddr_in *address)
+{
+	memcpy(&instance->local_address, address, sizeof(instance->local_address));
+	return 0;
+}
+
+int ast_rtp_instance_set_remote_address(struct ast_rtp_instance *instance, struct sockaddr_in *address)
+{
+	if (&instance->remote_address != address) {
+		memcpy(&instance->remote_address, address, sizeof(instance->remote_address));
+	}
+
+	/* moo */
+
+	if (instance->engine->remote_address_set) {
+		instance->engine->remote_address_set(instance, address);
+	}
+
+	return 0;
+}
+
+int ast_rtp_instance_get_local_address(struct ast_rtp_instance *instance, struct sockaddr_in *address)
+{
+	if ((address->sin_family != AF_INET) ||
+	    (address->sin_port != instance->local_address.sin_port) ||
+	    (address->sin_addr.s_addr != instance->local_address.sin_addr.s_addr)) {
+		memcpy(address, &instance->local_address, sizeof(address));
+		return 1;
+	}
+
+	return 0;
+}
+
+int ast_rtp_instance_get_remote_address(struct ast_rtp_instance *instance, struct sockaddr_in *address)
+{
+	if ((address->sin_family != AF_INET) ||
+	    (address->sin_port != instance->remote_address.sin_port) ||
+	    (address->sin_addr.s_addr != instance->remote_address.sin_addr.s_addr)) {
+		memcpy(address, &instance->remote_address, sizeof(address));
+		return 1;
+	}
+
+	return 0;
+}
+
+void ast_rtp_instance_set_extended_prop(struct ast_rtp_instance *instance, int property, void *value)
+{
+	if (instance->engine->extended_prop_set) {
+		instance->engine->extended_prop_set(instance, property, value);
+	}
+}
+
+void *ast_rtp_instance_get_extended_prop(struct ast_rtp_instance *instance, int property)
+{
+	if (instance->engine->extended_prop_get) {
+		return instance->engine->extended_prop_get(instance, property);
+	}
+
+	return NULL;
+}
+
+void ast_rtp_instance_set_prop(struct ast_rtp_instance *instance, enum ast_rtp_property property, int value)
+{
+	instance->properties[property] = value;
+
+	if (instance->engine->prop_set) {
+		instance->engine->prop_set(instance, property, value);
+	}
+}
+
+int ast_rtp_instance_get_prop(struct ast_rtp_instance *instance, enum ast_rtp_property property)
+{
+	return instance->properties[property];
+}
+
+struct ast_rtp_codecs *ast_rtp_instance_get_codecs(struct ast_rtp_instance *instance)
+{
+	return &instance->codecs;
+}
+
+void ast_rtp_codecs_payloads_clear(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance)
+{
+	int i;
+
+	for (i = 0; i < AST_RTP_MAX_PT; i++) {
+		ast_debug(2, "Clearing payload %d on %p\n", i, codecs);
+		codecs->payloads[i].asterisk_format = 0;
+		codecs->payloads[i].code = 0;
+		if (instance && instance->engine && instance->engine->payload_set) {
+			instance->engine->payload_set(instance, i, 0, 0);
+		}
+	}
+}
+
+void ast_rtp_codecs_payloads_default(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance)
+{
+	int i;
+
+	for (i = 0; i < AST_RTP_MAX_PT; i++) {
+		if (static_RTP_PT[i].code) {
+			ast_debug(2, "Set default payload %d on %p\n", i, codecs);
+			codecs->payloads[i].asterisk_format = static_RTP_PT[i].asterisk_format;
+			codecs->payloads[i].code = static_RTP_PT[i].code;
+			if (instance && instance->engine && instance->engine->payload_set) {
+				instance->engine->payload_set(instance, i, codecs->payloads[i].asterisk_format, codecs->payloads[i].code);
+			}
+		}
+	}
+}
+
+void ast_rtp_codecs_payloads_copy(struct ast_rtp_codecs *src, struct ast_rtp_codecs *dest, struct ast_rtp_instance *instance)
+{
+	int i;
+
+	for (i = 0; i < AST_RTP_MAX_PT; i++) {
+		if (src->payloads[i].code) {
+			ast_debug(2, "Copying payload %d from %p to %p\n", i, src, dest);
+			dest->payloads[i].asterisk_format = src->payloads[i].asterisk_format;
+			dest->payloads[i].code = src->payloads[i].code;
+			if (instance && instance->engine && instance->engine->payload_set) {
+				instance->engine->payload_set(instance, i, dest->payloads[i].asterisk_format, dest->payloads[i].code);
+			}
+		}
+	}
+}
+
+void ast_rtp_codecs_payloads_set_m_type(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload)
+{
+	if (payload < 0 || payload > AST_RTP_MAX_PT || !static_RTP_PT[payload].code) {
+		return;
+	}
+
+	codecs->payloads[payload].asterisk_format = static_RTP_PT[payload].asterisk_format;
+	codecs->payloads[payload].code = static_RTP_PT[payload].code;
+
+	ast_debug(1, "Setting payload %d based on m type on %p\n", payload, codecs);
+
+	if (instance && instance->engine && instance->engine->payload_set) {
+		instance->engine->payload_set(instance, payload, codecs->payloads[payload].asterisk_format, codecs->payloads[payload].code);
+	}
+}
+
+int ast_rtp_codecs_payloads_set_rtpmap_type_rate(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int pt,
+				 char *mimetype, char *mimesubtype,
+				 enum ast_rtp_options options,
+				 unsigned int sample_rate)
+{
+	unsigned int i;
+	int found = 0;
+
+	if (pt < 0 || pt > AST_RTP_MAX_PT)
+		return -1; /* bogus payload type */
+
+	for (i = 0; i < ARRAY_LEN(ast_rtp_mime_types); ++i) {
+		const struct ast_rtp_mime_type *t = &ast_rtp_mime_types[i];
+
+		if (strcasecmp(mimesubtype, t->subtype)) {
+			continue;
+		}
+
+		if (strcasecmp(mimetype, t->type)) {
+			continue;
+		}
+
+		/* if both sample rates have been supplied, and they don't match,
+		                      then this not a match; if one has not been supplied, then the
+				      rates are not compared */
+		if (sample_rate && t->sample_rate &&
+		    (sample_rate != t->sample_rate)) {
+			continue;
+		}
+
+		found = 1;
+		codecs->payloads[pt] = t->payload_type;
+
+		if ((t->payload_type.code == AST_FORMAT_G726) &&
+		                        t->payload_type.asterisk_format &&
+		    (options & AST_RTP_OPT_G726_NONSTANDARD)) {
+			codecs->payloads[pt].code = AST_FORMAT_G726_AAL2;
+		}
+
+		if (instance && instance->engine && instance->engine->payload_set) {
+			instance->engine->payload_set(instance, pt, codecs->payloads[i].asterisk_format, codecs->payloads[i].code);
+		}
+
+		break;
+	}
+
+	return (found ? 0 : -2);
+}
+
+int ast_rtp_codecs_payloads_set_rtpmap_type(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload, char *mimetype, char *mimesubtype, enum ast_rtp_options options)
+{
+	return ast_rtp_codecs_payloads_set_rtpmap_type_rate(codecs, instance, payload, mimetype, mimesubtype, options, 0);
+}
+
+void ast_rtp_codecs_payloads_unset(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload)
+{
+	if (payload < 0 || payload > AST_RTP_MAX_PT) {
+		return;
+	}
+
+	ast_debug(2, "Unsetting payload %d on %p\n", payload, codecs);
+
+	codecs->payloads[payload].asterisk_format = 0;
+	codecs->payloads[payload].code = 0;
+
+	if (instance && instance->engine && instance->engine->payload_set) {
+		instance->engine->payload_set(instance, payload, 0, 0);
+	}
+}
+
+struct ast_rtp_payload_type ast_rtp_codecs_payload_lookup(struct ast_rtp_codecs *codecs, int payload)
+{
+	struct ast_rtp_payload_type result = { .asterisk_format = 0, };
+
+	if (payload < 0 || payload > AST_RTP_MAX_PT) {
+		return result;
+	}
+
+	result.asterisk_format = codecs->payloads[payload].asterisk_format;
+	result.code = codecs->payloads[payload].code;
+
+	if (!result.code) {
+		result = static_RTP_PT[payload];
+	}
+
+	return result;
+}
+
+void ast_rtp_codecs_payload_formats(struct ast_rtp_codecs *codecs, int *astformats, int *nonastformats)
+{
+	int i;
+
+	*astformats = *nonastformats = 0;
+
+	for (i = 0; i < AST_RTP_MAX_PT; i++) {
+		if (codecs->payloads[i].code) {
+			ast_debug(1, "Incorporating payload %d on %p\n", i, codecs);
+		}
+		if (codecs->payloads[i].asterisk_format) {
+			*astformats |= codecs->payloads[i].code;
+		} else {
+			*nonastformats |= codecs->payloads[i].code;
+		}
+	}
+}
+
+int ast_rtp_codecs_payload_code(struct ast_rtp_codecs *codecs, const int asterisk_format, const int code)
+{
+	int i;
+
+	for (i = 0; i < AST_RTP_MAX_PT; i++) {
+		if (codecs->payloads[i].asterisk_format == asterisk_format && codecs->payloads[i].code == code) {
+			ast_debug(2, "Found code %d at payload %d on %p\n", code, i, codecs);
+			return i;
+		}
+	}
+
+	for (i = 0; i < AST_RTP_MAX_PT; i++) {
+		if (static_RTP_PT[i].asterisk_format == asterisk_format && static_RTP_PT[i].code == code) {
+			return i;
+		}
+	}
+
+	return -1;
+}
+
+const char *ast_rtp_lookup_mime_subtype2(const int asterisk_format, const int code, enum ast_rtp_options options)
+{
+	int i;
+
+	for (i = 0; i < ARRAY_LEN(ast_rtp_mime_types); i++) {
+		if (ast_rtp_mime_types[i].payload_type.code == code && ast_rtp_mime_types[i].payload_type.asterisk_format == asterisk_format) {
+			if (asterisk_format && (code == AST_FORMAT_G726_AAL2) && (options & AST_RTP_OPT_G726_NONSTANDARD)) {
+				return "G726-32";
+			} else {
+				return ast_rtp_mime_types[i].subtype;
+			}
+		}
+	}
+
+	return "";
+}
+
+unsigned int ast_rtp_lookup_sample_rate2(int asterisk_format, int code)
+{
+	unsigned int i;
+
+	for (i = 0; i < ARRAY_LEN(ast_rtp_mime_types); ++i) {
+		if ((ast_rtp_mime_types[i].payload_type.code == code) && (ast_rtp_mime_types[i].payload_type.asterisk_format == asterisk_format)) {
+			return ast_rtp_mime_types[i].sample_rate;
+		}
+	}
+
+	return 0;
+}
+
+char *ast_rtp_lookup_mime_multiple2(struct ast_str *buf, const int capability, const int asterisk_format, enum ast_rtp_options options)
+{
+	int format, found = 0;
+
+	if (!buf) {
+		return NULL;
+	}
+
+	ast_str_append(&buf, 0, "0x%x (", capability);
+
+	for (format = 1; format < AST_RTP_MAX; format <<= 1) {
+		if (capability & format) {
+			const char *name = ast_rtp_lookup_mime_subtype2(asterisk_format, format, options);
+			ast_str_append(&buf, 0, "%s|", name);
+			found = 1;
+		}
+	}
+
+	ast_str_append(&buf, 0, "%s", found ? ")" : "nothing)");
+
+	return ast_str_buffer(buf);
+}
+
+void ast_rtp_codecs_packetization_set(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, struct ast_codec_pref *prefs)
+{
+	codecs->pref = *prefs;
+
+	if (instance && instance->engine->packetization_set) {
+		instance->engine->packetization_set(instance, &instance->codecs.pref);
+	}
+}
+
+int ast_rtp_instance_dtmf_begin(struct ast_rtp_instance *instance, char digit)
+{
+	return instance->engine->dtmf_begin ? instance->engine->dtmf_begin(instance, digit) : -1;
+}
+
+int ast_rtp_instance_dtmf_end(struct ast_rtp_instance *instance, char digit)
+{
+	return instance->engine->dtmf_end ? instance->engine->dtmf_end(instance, digit) : -1;
+}
+
+int ast_rtp_instance_dtmf_mode_set(struct ast_rtp_instance *instance, enum ast_rtp_dtmf_mode dtmf_mode)
+{
+	if (!instance->engine->dtmf_mode_set || instance->engine->dtmf_mode_set(instance, dtmf_mode)) {
+		return -1;
+	}
+
+	instance->dtmf_mode = dtmf_mode;
+
+	return 0;
+}
+
+enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get(struct ast_rtp_instance *instance)
+{
+	return instance->dtmf_mode;
+}
+
+void ast_rtp_instance_new_source(struct ast_rtp_instance *instance)
+{
+	if (instance->engine->new_source) {
+		instance->engine->new_source(instance);
+	}
+}
+
+int ast_rtp_instance_set_qos(struct ast_rtp_instance *instance, int tos, int cos, const char *desc)
+{
+	return instance->engine->qos ? instance->engine->qos(instance, tos, cos, desc) : -1;
+}
+
+void ast_rtp_instance_stop(struct ast_rtp_instance *instance)
+{
+	if (instance->engine->stop) {
+		instance->engine->stop(instance);
+	}
+}
+
+int ast_rtp_instance_fd(struct ast_rtp_instance *instance, int rtcp)
+{
+	return instance->engine->fd ? instance->engine->fd(instance, rtcp) : -1;
+}
+
+struct ast_rtp_glue *ast_rtp_instance_get_glue(const char *type)
+{
+	struct ast_rtp_glue *glue = NULL;
+
+	AST_RWLIST_RDLOCK(&glues);
+
+	AST_RWLIST_TRAVERSE(&glues, glue, entry) {
+		if (!strcasecmp(glue->type, type)) {
+			break;
+		}
+	}
+
+	AST_RWLIST_UNLOCK(&glues);
+
+	return glue;
+}
+
+static enum ast_bridge_result local_bridge_loop(struct ast_channel *c0, struct ast_channel *c1, struct ast_rtp_instance *instance0, struct ast_rtp_instance *instance1, int timeoutms, int flags, struct ast_frame **fo, struct ast_channel **rc, void *pvt0, void *pvt1)
+{
+	enum ast_bridge_result res = AST_BRIDGE_FAILED;
+	struct ast_channel *who = NULL, *other = NULL, *cs[3] = { NULL, };
+	struct ast_frame *fr = NULL;
+
+	/* Start locally bridging both instances */
+	if (instance0->engine->local_bridge && instance0->engine->local_bridge(instance0, instance1)) {
+		ast_debug(1, "Failed to locally bridge %s to %s, backing out.\n", c0->name, c1->name);
+		ast_channel_unlock(c0);
+		ast_channel_unlock(c1);
+		return AST_BRIDGE_FAILED_NOWARN;
+	}
+	if (instance1->engine->local_bridge && instance1->engine->local_bridge(instance1, instance0)) {
+		ast_debug(1, "Failed to locally bridge %s to %s, backing out.\n", c1->name, c0->name);
+		if (instance0->engine->local_bridge) {
+			instance0->engine->local_bridge(instance0, NULL);
+		}
+		ast_channel_unlock(c0);
+		ast_channel_unlock(c1);
+		return AST_BRIDGE_FAILED_NOWARN;
+	}
+
+	ast_channel_unlock(c0);
+	ast_channel_unlock(c1);
+
+	instance0->bridged = instance1;
+	instance1->bridged = instance0;
+
+	ast_poll_channel_add(c0, c1);
+
+	/* Hop into a loop waiting for a frame from either channel */
+	cs[0] = c0;
+	cs[1] = c1;
+	cs[2] = NULL;
+	for (;;) {
+		/* If the underlying formats have changed force this bridge to break */
+		if ((c0->rawreadformat != c1->rawwriteformat) || (c1->rawreadformat != c0->rawwriteformat)) {
+			ast_debug(1, "rtp-engine-local-bridge: Oooh, formats changed, backing out\n");
+			res = AST_BRIDGE_FAILED_NOWARN;
+			break;
+		}
+		/* Check if anything changed */
+		if ((c0->tech_pvt != pvt0) ||
+		    (c1->tech_pvt != pvt1) ||
+		    (c0->masq || c0->masqr || c1->masq || c1->masqr) ||
+		    (c0->monitor || c0->audiohooks || c1->monitor || c1->audiohooks)) {
+			ast_debug(1, "rtp-engine-local-bridge: Oooh, something is weird, backing out\n");
+			/* If a masquerade needs to happen we have to try to read in a frame so that it actually happens. Without this we risk being called again and going into a loop */
+			if ((c0->masq || c0->masqr) && (fr = ast_read(c0))) {
+				ast_frfree(fr);
+			}
+			if ((c1->masq || c1->masqr) && (fr = ast_read(c1))) {
+				ast_frfree(fr);
+			}
+			res = AST_BRIDGE_RETRY;
+			break;
+		}
+		/* Wait on a channel to feed us a frame */
+		if (!(who = ast_waitfor_n(cs, 2, &timeoutms))) {
+			if (!timeoutms) {
+				res = AST_BRIDGE_RETRY;
+				break;
+			}
+			ast_debug(2, "rtp-engine-local-bridge: Ooh, empty read...\n");
+			if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
+				break;
+			}
+			continue;
+		}
+		/* Read in frame from channel */
+		fr = ast_read(who);
+		other = (who == c0) ? c1 : c0;
+		/* Depending on the frame we may need to break out of our bridge */
+		if (!fr || ((fr->frametype == AST_FRAME_DTMF_BEGIN || fr->frametype == AST_FRAME_DTMF_END) &&
+			    ((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) |
+			    ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)))) {
+			/* Record received frame and who */
+			*fo = fr;
+			*rc = who;
+			ast_debug(1, "rtp-engine-local-bridge: Ooh, got a %s\n", fr ? "digit" : "hangup");
+			res = AST_BRIDGE_COMPLETE;
+			break;
+		} else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
+			if ((fr->subclass == AST_CONTROL_HOLD) ||
+			    (fr->subclass == AST_CONTROL_UNHOLD) ||
+			    (fr->subclass == AST_CONTROL_VIDUPDATE) ||
+			    (fr->subclass == AST_CONTROL_T38) ||
+			    (fr->subclass == AST_CONTROL_SRCUPDATE)) {
+				/* If we are going on hold, then break callback mode and P2P bridging */
+				if (fr->subclass == AST_CONTROL_HOLD) {
+					if (instance0->engine->local_bridge) {
+						instance0->engine->local_bridge(instance0, NULL);
+					}
+					if (instance1->engine->local_bridge) {
+						instance1->engine->local_bridge(instance1, NULL);
+					}
+					instance0->bridged = NULL;
+					instance1->bridged = NULL;
+				} else if (fr->subclass == AST_CONTROL_UNHOLD) {
+					if (instance0->engine->local_bridge) {
+						instance0->engine->local_bridge(instance0, instance1);
+					}
+					if (instance1->engine->local_bridge) {
+						instance1->engine->local_bridge(instance1, instance0);
+					}
+					instance0->bridged = instance1;
+					instance1->bridged = instance0;
+				}
+				ast_indicate_data(other, fr->subclass, fr->data.ptr, fr->datalen);
+				ast_frfree(fr);
+			} else {
+				*fo = fr;
+				*rc = who;
+				ast_debug(1, "rtp-engine-local-bridge: Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass, who->name);
+				res = AST_BRIDGE_COMPLETE;
+				break;
+			}
+		} else {
+			if ((fr->frametype == AST_FRAME_DTMF_BEGIN) ||
+			    (fr->frametype == AST_FRAME_DTMF_END) ||
+			    (fr->frametype == AST_FRAME_VOICE) ||
+			    (fr->frametype == AST_FRAME_VIDEO) ||
+			    (fr->frametype == AST_FRAME_IMAGE) ||
+			    (fr->frametype == AST_FRAME_HTML) ||
+			    (fr->frametype == AST_FRAME_MODEM) ||
+			    (fr->frametype == AST_FRAME_TEXT)) {
+				ast_write(other, fr);
+			}
+
+			ast_frfree(fr);
+		}
+		/* Swap priority */
+		cs[2] = cs[0];
+		cs[0] = cs[1];
+		cs[1] = cs[2];
+	}
+
+	/* Stop locally bridging both instances */
+	if (instance0->engine->local_bridge) {
+		instance0->engine->local_bridge(instance0, NULL);
+	}
+	if (instance1->engine->local_bridge) {
+		instance1->engine->local_bridge(instance1, NULL);
+	}
+
+	instance0->bridged = NULL;
+	instance1->bridged = NULL;
+
+	ast_poll_channel_del(c0, c1);
+
+	return res;
+}
+
+static enum ast_bridge_result remote_bridge_loop(struct ast_channel *c0, struct ast_channel *c1, struct ast_rtp_instance *instance0, struct ast_rtp_instance *instance1,
+						 struct ast_rtp_instance *vinstance0, struct ast_rtp_instance *vinstance1, struct ast_rtp_instance *tinstance0,
+						 struct ast_rtp_instance *tinstance1, struct ast_rtp_glue *glue0, struct ast_rtp_glue *glue1, int codec0, int codec1, int timeoutms,
+						 int flags, struct ast_frame **fo, struct ast_channel **rc, void *pvt0, void *pvt1)
+{
+	enum ast_bridge_result res = AST_BRIDGE_FAILED;
+	struct ast_channel *who = NULL, *other = NULL, *cs[3] = { NULL, };
+	int oldcodec0 = codec0, oldcodec1 = codec1;
+	struct sockaddr_in ac1 = {0,}, vac1 = {0,}, tac1 = {0,}, ac0 = {0,}, vac0 = {0,}, tac0 = {0,};
+	struct sockaddr_in t1 = {0,}, vt1 = {0,}, tt1 = {0,}, t0 = {0,}, vt0 = {0,}, tt0 = {0,};
+	struct ast_frame *fr = NULL;
+
+	/* Test the first channel */
+	if (!(glue0->update_peer(c0, instance1, vinstance1, tinstance1, codec1, 0))) {
+		ast_rtp_instance_get_remote_address(instance1, &ac1);
+		if (vinstance1) {
+			ast_rtp_instance_get_remote_address(vinstance1, &vac1);
+		}
+		if (tinstance1) {
+			ast_rtp_instance_get_remote_address(tinstance1, &tac1);
+		}
+	} else {
+		ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c0->name, c1->name);
+	}
+
+	/* Test the second channel */
+	if (!(glue1->update_peer(c1, instance0, vinstance0, tinstance0, codec0, 0))) {
+		ast_rtp_instance_get_remote_address(instance0, &ac0);
+		if (vinstance0) {
+			ast_rtp_instance_get_remote_address(instance0, &vac0);
+		}
+		if (tinstance0) {
+			ast_rtp_instance_get_remote_address(instance0, &tac0);
+		}
+	} else {
+		ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c1->name, c0->name);
+	}
+
+	ast_channel_unlock(c0);
+	ast_channel_unlock(c1);
+
+	instance0->bridged = instance1;
+	instance1->bridged = instance0;
+
+	ast_poll_channel_add(c0, c1);
+
+	/* Go into a loop handling any stray frames that may come in */
+	cs[0] = c0;
+	cs[1] = c1;
+	cs[2] = NULL;
+	for (;;) {
+		/* Check if anything changed */
+		if ((c0->tech_pvt != pvt0) ||
+		    (c1->tech_pvt != pvt1) ||
+		    (c0->masq || c0->masqr || c1->masq || c1->masqr) ||
+		    (c0->monitor || c0->audiohooks || c1->monitor || c1->audiohooks)) {
+			ast_debug(1, "Oooh, something is weird, backing out\n");
+			res = AST_BRIDGE_RETRY;
+			break;
+		}
+
+		/* Check if they have changed their address */
+		ast_rtp_instance_get_remote_address(instance1, &t1);
+		if (vinstance1) {
+			ast_rtp_instance_get_remote_address(vinstance1, &vt1);
+		}
+		if (tinstance1) {
+			ast_rtp_instance_get_remote_address(tinstance1, &tt1);
+		}
+		if (glue1->get_codec) {
+			codec1 = glue1->get_codec(c1);
+		}
+
+		ast_rtp_instance_get_remote_address(instance0, &t0);
+		if (vinstance0) {
+			ast_rtp_instance_get_remote_address(vinstance0, &vt0);
+		}
+		if (tinstance0) {
+			ast_rtp_instance_get_remote_address(tinstance0, &tt0);
+		}
+		if (glue0->get_codec) {
+			codec0 = glue0->get_codec(c0);
+		}
+
+		if ((inaddrcmp(&t1, &ac1)) ||
+		    (vinstance1 && inaddrcmp(&vt1, &vac1)) ||
+		    (tinstance1 && inaddrcmp(&tt1, &tac1)) ||
+		    (codec1 != oldcodec1)) {
+			ast_debug(1, "Oooh, '%s' changed end address to %s:%d (format %d)\n",
+				  c1->name, ast_inet_ntoa(t1.sin_addr), ntohs(t1.sin_port), codec1);
+			ast_debug(1, "Oooh, '%s' changed end vaddress to %s:%d (format %d)\n",
+				  c1->name, ast_inet_ntoa(vt1.sin_addr), ntohs(vt1.sin_port), codec1);
+			ast_debug(1, "Oooh, '%s' changed end taddress to %s:%d (format %d)\n",
+				  c1->name, ast_inet_ntoa(tt1.sin_addr), ntohs(tt1.sin_port), codec1);
+			ast_debug(1, "Oooh, '%s' was %s:%d/(format %d)\n",
+				  c1->name, ast_inet_ntoa(ac1.sin_addr), ntohs(ac1.sin_port), oldcodec1);
+			ast_debug(1, "Oooh, '%s' was %s:%d/(format %d)\n",
+				  c1->name, ast_inet_ntoa(vac1.sin_addr), ntohs(vac1.sin_port), oldcodec1);
+			ast_debug(1, "Oooh, '%s' was %s:%d/(format %d)\n",
+				  c1->name, ast_inet_ntoa(tac1.sin_addr), ntohs(tac1.sin_port), oldcodec1);
+			if (glue0->update_peer(c0, t1.sin_addr.s_addr ? instance1 : NULL, vt1.sin_addr.s_addr ? vinstance1 : NULL, tt1.sin_addr.s_addr ? tinstance1 : NULL, codec1, 0)) {
+				ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c0->name, c1->name);
+			}
+			memcpy(&ac1, &t1, sizeof(ac1));
+			memcpy(&vac1, &vt1, sizeof(vac1));
+			memcpy(&tac1, &tt1, sizeof(tac1));
+			oldcodec1 = codec1;
+		}
+		if ((inaddrcmp(&t0, &ac0)) ||
+		    (vinstance0 && inaddrcmp(&vt0, &vac0)) ||
+		    (tinstance0 && inaddrcmp(&tt0, &tac0))) {
+			ast_debug(1, "Oooh, '%s' changed end address to %s:%d (format %d)\n",
+				  c0->name, ast_inet_ntoa(t0.sin_addr), ntohs(t0.sin_port), codec0);
+			ast_debug(1, "Oooh, '%s' was %s:%d/(format %d)\n",
+				  c0->name, ast_inet_ntoa(ac0.sin_addr), ntohs(ac0.sin_port), oldcodec0);
+			if (glue1->update_peer(c1, t0.sin_addr.s_addr ? instance0 : NULL, vt0.sin_addr.s_addr ? vinstance0 : NULL, tt0.sin_addr.s_addr ? tinstance0 : NULL, codec0, 0)) {
+				ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c1->name, c0->name);
+			}
+			memcpy(&ac0, &t0, sizeof(ac0));
+			memcpy(&vac0, &vt0, sizeof(vac0));
+			memcpy(&tac0, &tt0, sizeof(tac0));
+			oldcodec0 = codec0;
+		}
+
+		/* Wait for frame to come in on the channels */
+		if (!(who = ast_waitfor_n(cs, 2, &timeoutms))) {
+			if (!timeoutms) {
+				res = AST_BRIDGE_RETRY;
+				break;
+			}
+			ast_debug(1, "Ooh, empty read...\n");
+			if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
+				break;
+			}
+			continue;
+		}
+		fr = ast_read(who);
+		other = (who == c0) ? c1 : c0;
+		if (!fr || ((fr->frametype == AST_FRAME_DTMF_BEGIN || fr->frametype == AST_FRAME_DTMF_END) &&
+			    (((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) ||
+			     ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1))))) {
+			/* Break out of bridge */
+			*fo = fr;
+			*rc = who;
+			ast_debug(1, "Oooh, got a %s\n", fr ? "digit" : "hangup");
+			res = AST_BRIDGE_COMPLETE;
+			break;
+		} else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
+			if ((fr->subclass == AST_CONTROL_HOLD) ||
+			    (fr->subclass == AST_CONTROL_UNHOLD) ||
+			    (fr->subclass == AST_CONTROL_VIDUPDATE) ||
+			    (fr->subclass == AST_CONTROL_T38) ||
+			    (fr->subclass == AST_CONTROL_SRCUPDATE)) {
+				if (fr->subclass == AST_CONTROL_HOLD) {
+					/* If we someone went on hold we want the other side to reinvite back to us */
+					if (who == c0) {
+						glue1->update_peer(c1, NULL, NULL, NULL, 0, 0);
+					} else {
+						glue0->update_peer(c0, NULL, NULL, NULL, 0, 0);
+					}
+				} else if (fr->subclass == AST_CONTROL_UNHOLD) {
+					/* If they went off hold they should go back to being direct */
+					if (who == c0) {
+						glue1->update_peer(c1, instance0, vinstance0, tinstance0, codec0, 0);
+					} else {
+						glue0->update_peer(c0, instance1, vinstance1, tinstance1, codec1, 0);
+					}
+				}
+				/* Update local address information */
+				ast_rtp_instance_get_remote_address(instance0, &t0);
+				memcpy(&ac0, &t0, sizeof(ac0));
+				ast_rtp_instance_get_remote_address(instance1, &t1);
+				memcpy(&ac1, &t1, sizeof(ac1));
+				/* Update codec information */
+				if (glue0->get_codec && c0->tech_pvt) {
+					oldcodec0 = codec0 = glue0->get_codec(c0);
+				}
+				if (glue1->get_codec && c1->tech_pvt) {
+					oldcodec1 = codec1 = glue1->get_codec(c1);
+				}
+				ast_indicate_data(other, fr->subclass, fr->data.ptr, fr->datalen);
+				ast_frfree(fr);
+			} else {
+				*fo = fr;
+				*rc = who;
+				ast_debug(1, "Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass, who->name);
+				return AST_BRIDGE_COMPLETE;
+			}
+		} else {
+			if ((fr->frametype == AST_FRAME_DTMF_BEGIN) ||
+			    (fr->frametype == AST_FRAME_DTMF_END) ||
+			    (fr->frametype == AST_FRAME_VOICE) ||
+			    (fr->frametype == AST_FRAME_VIDEO) ||
+			    (fr->frametype == AST_FRAME_IMAGE) ||
+			    (fr->frametype == AST_FRAME_HTML) ||
+			    (fr->frametype == AST_FRAME_MODEM) ||
+			    (fr->frametype == AST_FRAME_TEXT)) {
+				ast_write(other, fr);
+			}
+			ast_frfree(fr);
+		}
+		/* Swap priority */
+		cs[2] = cs[0];
+		cs[0] = cs[1];
+		cs[1] = cs[2];
+	}
+
+	if (glue0->update_peer(c0, NULL, NULL, NULL, 0, 0)) {
+		ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
+	}
+	if (glue1->update_peer(c1, NULL, NULL, NULL, 0, 0)) {
+		ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
+	}
+
+	instance0->bridged = NULL;
+	instance1->bridged = NULL;
+
+	ast_poll_channel_del(c0, c1);
+
+	return res;
+}
+
+/*!
+ * \brief Conditionally unref an rtp instance
+ */
+static void unref_instance_cond(struct ast_rtp_instance **instance)
+{
+	if (*instance) {
+		ao2_ref(*instance, -1);
+		*instance = NULL;
+	}
+}
+
+enum ast_bridge_result ast_rtp_instance_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms)
+{
+	struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
+			*vinstance0 = NULL, *vinstance1 = NULL,
+			*tinstance0 = NULL, *tinstance1 = NULL;
+	struct ast_rtp_glue *glue0, *glue1;
+	enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID, text_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
+	enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID, text_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
+	enum ast_bridge_result res = AST_BRIDGE_FAILED;
+	int codec0 = 0, codec1 = 0;
+	int unlock_chans = 1;
+
+	/* Lock both channels so we can look for the glue that binds them together */
+	ast_channel_lock(c0);
+	while (ast_channel_trylock(c1)) {
+		ast_channel_unlock(c0);
+		usleep(1);
+		ast_channel_lock(c0);
+	}
+
+	/* Ensure neither channel got hungup during lock avoidance */
+	if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
+		ast_log(LOG_WARNING, "Got hangup while attempting to bridge '%s' and '%s'\n", c0->name, c1->name);
+		goto done;
+	}
+
+	/* Grab glue that binds each channel to something using the RTP engine */
+	if (!(glue0 = ast_rtp_instance_get_glue(c0->tech->type)) || !(glue1 = ast_rtp_instance_get_glue(c1->tech->type))) {
+		ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", glue0 ? c1->name : c0->name);
+		goto done;
+	}
+
+	audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
+	video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
+	text_glue0_res = glue0->get_trtp_info ? glue0->get_trtp_info(c0, &tinstance0) : AST_RTP_GLUE_RESULT_FORBID;
+
+	audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
+	video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
+	text_glue1_res = glue1->get_trtp_info ? glue1->get_trtp_info(c1, &tinstance1) : AST_RTP_GLUE_RESULT_FORBID;
+
+	/* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
+	if (video_glue0_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue0_res != AST_RTP_GLUE_RESULT_REMOTE)) {
+		audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
+	}
+	if (video_glue1_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue1_res != AST_RTP_GLUE_RESULT_REMOTE)) {
+		audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
+	}
+
+	/* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
+	if (audio_glue0_res == AST_RTP_GLUE_RESULT_FORBID || audio_glue1_res == AST_RTP_GLUE_RESULT_FORBID) {
+		res = AST_BRIDGE_FAILED_NOWARN;
+		goto done;
+	}
+
+	/* If we have gotten to a local bridge make sure that both sides have the same local bridge callback and that they are DTMF compatible */
+	if ((audio_glue0_res == AST_RTP_GLUE_RESULT_LOCAL || audio_glue1_res == AST_RTP_GLUE_RESULT_LOCAL) && ((instance0->engine->local_bridge != instance1->engine->local_bridge) || (instance0->engine->dtmf_compatible && !instance0->engine->dtmf_compatible(c0, instance0, c1, instance1)))) {
+		res = AST_BRIDGE_FAILED_NOWARN;
+		goto done;
+	}
+
+	/* Make sure that codecs match */
+	codec0 = glue0->get_codec ? glue0->get_codec(c0) : 0;
+	codec1 = glue1->get_codec ? glue1->get_codec(c1) : 0;
+	if (codec0 && codec1 && !(codec0 & codec1)) {
+		ast_debug(1, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1);
+		res = AST_BRIDGE_FAILED_NOWARN;
+		goto done;
+	}
+
+	/* Depending on the end result for bridging either do a local bridge or remote bridge */
+	if (audio_glue0_res == AST_RTP_GLUE_RESULT_LOCAL || audio_glue1_res == AST_RTP_GLUE_RESULT_LOCAL) {
+		ast_verbose(VERBOSE_PREFIX_3 "Locally bridging %s and %s\n", c0->name, c1->name);
+		res = local_bridge_loop(c0, c1, instance0, instance1, timeoutms, flags, fo, rc, c0->tech_pvt, c1->tech_pvt);
+	} else {
+		ast_verbose(VERBOSE_PREFIX_3 "Remotely bridging %s and %s\n", c0->name, c1->name);
+		res = remote_bridge_loop(c0, c1, instance0, instance1, vinstance0, vinstance1,
+				tinstance0, tinstance1, glue0, glue1, codec0, codec1, timeoutms, flags,
+				fo, rc, c0->tech_pvt, c1->tech_pvt);
+	}
+
+	unlock_chans = 0;
+
+done:
+	if (unlock_chans) {
+		ast_channel_unlock(c0);
+		ast_channel_unlock(c1);
+	}
+
+	unref_instance_cond(&instance0);
+	unref_instance_cond(&instance1);
+	unref_instance_cond(&vinstance0);
+	unref_instance_cond(&vinstance1);
+	unref_instance_cond(&tinstance0);
+	unref_instance_cond(&tinstance1);
+
+	return res;
+}
+
+struct ast_rtp_instance *ast_rtp_instance_get_bridged(struct ast_rtp_instance *instance)
+{
+	return instance->bridged;
+}
+
+void ast_rtp_instance_early_bridge_make_compatible(struct ast_channel *c0, struct ast_channel *c1)
+{
+	struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
+		*vinstance0 = NULL, *vinstance1 = NULL,
+		*tinstance0 = NULL, *tinstance1 = NULL;
+	struct ast_rtp_glue *glue0, *glue1;
+	enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID, text_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
+	enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID, text_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
+	int codec0 = 0, codec1 = 0;
+	int res = 0;
+
+	/* Lock both channels so we can look for the glue that binds them together */
+	ast_channel_lock(c0);
+	while (ast_channel_trylock(c1)) {
+		ast_channel_unlock(c0);
+		usleep(1);
+		ast_channel_lock(c0);
+	}
+
+	/* Grab glue that binds each channel to something using the RTP engine */
+	if (!(glue0 = ast_rtp_instance_get_glue(c0->tech->type)) || !(glue1 = ast_rtp_instance_get_glue(c1->tech->type))) {
+		ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", glue0 ? c1->name : c0->name);
+		goto done;
+	}
+
+	audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
+	video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
+	text_glue0_res = glue0->get_trtp_info ? glue0->get_trtp_info(c0, &tinstance0) : AST_RTP_GLUE_RESULT_FORBID;
+
+	audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
+	video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
+	text_glue1_res = glue1->get_trtp_info ? glue1->get_trtp_info(c1, &tinstance1) : AST_RTP_GLUE_RESULT_FORBID;
+
+	/* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
+	if (video_glue0_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue0_res != AST_RTP_GLUE_RESULT_REMOTE)) {
+		audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
+	}
+	if (video_glue1_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue1_res != AST_RTP_GLUE_RESULT_REMOTE)) {
+		audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
+	}
+	if (audio_glue0_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue0_res == AST_RTP_GLUE_RESULT_FORBID || video_glue0_res == AST_RTP_GLUE_RESULT_REMOTE) && glue0->get_codec(c0)) {
+		codec0 = glue0->get_codec(c0);
+	}
+	if (audio_glue1_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue1_res == AST_RTP_GLUE_RESULT_FORBID || video_glue1_res == AST_RTP_GLUE_RESULT_REMOTE) && glue1->get_codec(c1)) {
+		codec1 = glue1->get_codec(c1);
+	}
+
+	/* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
+	if (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE) {
+		goto done;
+	}
+
+	/* Make sure we have matching codecs */
+	if (!(codec0 & codec1)) {
+		goto done;
+	}
+
+	ast_rtp_codecs_payloads_copy(&instance0->codecs, &instance1->codecs, instance1);
+
+	if (vinstance0 && vinstance1) {
+		ast_rtp_codecs_payloads_copy(&vinstance0->codecs, &vinstance1->codecs, vinstance1);
+	}
+	if (tinstance0 && tinstance1) {
+		ast_rtp_codecs_payloads_copy(&tinstance0->codecs, &tinstance1->codecs, tinstance1);
+	}
+
+	res = 0;
+
+done:
+	ast_channel_unlock(c0);
+	ast_channel_unlock(c1);
+
+	unref_instance_cond(&instance0);
+	unref_instance_cond(&instance1);
+	unref_instance_cond(&vinstance0);
+	unref_instance_cond(&vinstance1);
+	unref_instance_cond(&tinstance0);
+	unref_instance_cond(&tinstance1);
+
+	if (!res) {
+		ast_debug(1, "Seeded SDP of '%s' with that of '%s'\n", c0->name, c1 ? c1->name : "<unspecified>");
+	}
+}
+
+int ast_rtp_instance_early_bridge(struct ast_channel *c0, struct ast_channel *c1)
+{
+	struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
+			*vinstance0 = NULL, *vinstance1 = NULL,
+			*tinstance0 = NULL, *tinstance1 = NULL;
+	struct ast_rtp_glue *glue0, *glue1;
+	enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID, text_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
+	enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID, text_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
+	int codec0 = 0, codec1 = 0;
+	int res = 0;
+
+	/* If there is no second channel just immediately bail out, we are of no use in that scenario */
+	if (!c1) {
+		return -1;
+	}
+
+	/* Lock both channels so we can look for the glue that binds them together */
+	ast_channel_lock(c0);
+	while (ast_channel_trylock(c1)) {
+		ast_channel_unlock(c0);
+		usleep(1);
+		ast_channel_lock(c0);
+	}
+
+	/* Grab glue that binds each channel to something using the RTP engine */
+	if (!(glue0 = ast_rtp_instance_get_glue(c0->tech->type)) || !(glue1 = ast_rtp_instance_get_glue(c1->tech->type))) {
+		ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", glue0 ? c1->name : c0->name);
+		goto done;
+	}
+
+	audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
+	video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
+	text_glue0_res = glue0->get_trtp_info ? glue0->get_trtp_info(c0, &tinstance0) : AST_RTP_GLUE_RESULT_FORBID;
+
+	audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
+	video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
+	text_glue1_res = glue1->get_trtp_info ? glue1->get_trtp_info(c1, &tinstance1) : AST_RTP_GLUE_RESULT_FORBID;
+
+	/* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
+	if (video_glue0_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue0_res != AST_RTP_GLUE_RESULT_REMOTE)) {
+		audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
+	}
+	if (video_glue1_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue1_res != AST_RTP_GLUE_RESULT_REMOTE)) {
+		audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
+	}
+	if (audio_glue0_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue0_res == AST_RTP_GLUE_RESULT_FORBID || video_glue0_res == AST_RTP_GLUE_RESULT_REMOTE) && glue0->get_codec(c0)) {
+		codec0 = glue0->get_codec(c0);
+	}
+	if (audio_glue1_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue1_res == AST_RTP_GLUE_RESULT_FORBID || video_glue1_res == AST_RTP_GLUE_RESULT_REMOTE) && glue1->get_codec(c1)) {
+		codec1 = glue1->get_codec(c1);
+	}
+
+	/* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
+	if (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE) {
+		goto done;
+	}
+
+	/* Make sure we have matching codecs */
+	if (!(codec0 & codec1)) {
+		goto done;
+	}
+
+	/* Bridge media early */
+	if (glue0->update_peer(c0, instance1, vinstance1, tinstance1, codec1, 0)) {
+		ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", c0->name, c1 ? c1->name : "<unspecified>");
+	}
+
+	res = 0;
+
+done:
+	ast_channel_unlock(c0);
+	ast_channel_unlock(c1);
+
+	unref_instance_cond(&instance0);
+	unref_instance_cond(&instance1);
+	unref_instance_cond(&vinstance0);
+	unref_instance_cond(&vinstance1);
+	unref_instance_cond(&tinstance0);
+	unref_instance_cond(&tinstance1);
+
+	if (!res) {
+		ast_debug(1, "Setting early bridge SDP of '%s' with that of '%s'\n", c0->name, c1 ? c1->name : "<unspecified>");
+	}
+
+	return res;
+}
+
+int ast_rtp_red_init(struct ast_rtp_instance *instance, int buffer_time, int *payloads, int generations)
+{
+	return instance->engine->red_init ? instance->engine->red_init(instance, buffer_time, payloads, generations) : -1;
+}
+
+int ast_rtp_red_buffer(struct ast_rtp_instance *instance, struct ast_frame *frame)
+{
+	return instance->engine->red_buffer ? instance->engine->red_buffer(instance, frame) : -1;
+}
+
+int ast_rtp_instance_get_stats(struct ast_rtp_instance *instance, struct ast_rtp_instance_stats *stats, enum ast_rtp_instance_stat stat)
+{
+	return instance->engine->get_stat ? instance->engine->get_stat(instance, stats, stat) : -1;
+}
+
+char *ast_rtp_instance_get_quality(struct ast_rtp_instance *instance, enum ast_rtp_instance_stat_field field, char *buf, size_t size)
+{
+	struct ast_rtp_instance_stats stats;
+	enum ast_rtp_instance_stat stat;
+
+	/* Determine what statistics we will need to retrieve based on field passed in */
+	if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY) {
+		stat = AST_RTP_INSTANCE_STAT_ALL;
+	} else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER) {
+		stat = AST_RTP_INSTANCE_STAT_COMBINED_JITTER;
+	} else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS) {
+		stat = AST_RTP_INSTANCE_STAT_COMBINED_LOSS;
+	} else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT) {
+		stat = AST_RTP_INSTANCE_STAT_COMBINED_RTT;
+	} else {
+		return NULL;
+	}
+
+	/* Attempt to actually retrieve the statistics we need to generate the quality string */
+	if (ast_rtp_instance_get_stats(instance, &stats, stat)) {
+		return NULL;
+	}
+
+	/* Now actually fill the buffer with the good information */
+	if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY) {
+		snprintf(buf, size, "ssrc=%i;themssrc=%u;lp=%u;rxjitter=%u;rxcount=%u;txjitter=%u;txcount=%u;rlp=%u;rtt=%u",
+			 stats.local_ssrc, stats.remote_ssrc, stats.rxploss, stats.txjitter, stats.rxcount, stats.rxjitter, stats.txcount, stats.txploss, stats.rtt);
+	} else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER) {
+		snprintf(buf, size, "minrxjitter=%f;maxrxjitter=%f;avgrxjitter=%f;stdevrxjitter=%f;reported_minjitter=%f;reported_maxjitter=%f;reported_avgjitter=%f;reported_stdevjitter=%f;",
+			 stats.local_minjitter, stats.local_maxjitter, stats.local_normdevjitter, sqrt(stats.local_stdevjitter), stats.remote_minjitter, stats.remote_maxjitter, stats.remote_normdevjitter, sqrt(stats.remote_stdevjitter));
+	} else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS) {
+		snprintf(buf, size, "minrxlost=%f;maxrxlost=%f;avgrxlost=%f;stdevrxlost=%f;reported_minlost=%f;reported_maxlost=%f;reported_avglost=%f;reported_stdevlost=%f;",
+			 stats.local_minrxploss, stats.local_maxrxploss, stats.local_normdevrxploss, sqrt(stats.local_stdevrxploss), stats.remote_minrxploss, stats.remote_maxrxploss, stats.remote_normdevrxploss, sqrt(stats.remote_stdevrxploss));
+	} else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT) {
+		snprintf(buf, size, "minrtt=%f;maxrtt=%f;avgrtt=%f;stdevrtt=%f;", stats.minrtt, stats.maxrtt, stats.normdevrtt, stats.stdevrtt);
+	}
+
+	return buf;
+}
+
+void ast_rtp_instance_set_stats_vars(struct ast_channel *chan, struct ast_rtp_instance *instance)
+{
+	char quality_buf[AST_MAX_USER_FIELD], *quality;
+	struct ast_channel *bridge = ast_bridged_channel(chan);
+
+	if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
+		pbx_builtin_setvar_helper(chan, "RTPAUDIOQOS", quality);
+		if (bridge) {
+			pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSBRIDGED", quality);
+		}
+	}
+
+	if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, quality_buf, sizeof(quality_buf)))) {
+		pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSJITTER", quality);
+		if (bridge) {
+			pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSJITTERBRIDGED", quality);
+		}
+	}
+
+	if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, quality_buf, sizeof(quality_buf)))) {
+		pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSLOSS", quality);
+		if (bridge) {
+			pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSLOSSBRIDGED", quality);
+		}
+	}
+
+	if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, quality_buf, sizeof(quality_buf)))) {
+		pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSRTT", quality);
+		if (bridge) {
+			pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSRTTBRIDGED", quality);
+		}
+	}
+}
+
+int ast_rtp_instance_set_read_format(struct ast_rtp_instance *instance, int format)
+{
+	return instance->engine->set_read_format ? instance->engine->set_read_format(instance, format) : -1;
+}
+
+int ast_rtp_instance_set_write_format(struct ast_rtp_instance *instance, int format)
+{
+	return instance->engine->set_write_format ? instance->engine->set_write_format(instance, format) : -1;
+}
+
+int ast_rtp_instance_make_compatible(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_channel *peer)
+{
+	struct ast_rtp_glue *glue;
+	struct ast_rtp_instance *peer_instance = NULL;
+	int res = -1;
+
+	if (!instance->engine->make_compatible) {
+		return -1;
+	}
+
+	ast_channel_lock(peer);
+
+	if (!(glue = ast_rtp_instance_get_glue(peer->tech->type))) {
+		ast_channel_unlock(peer);
+		return -1;
+	}
+
+	glue->get_rtp_info(peer, &peer_instance);
+
+	if (!peer_instance || peer_instance->engine != instance->engine) {
+		ast_channel_unlock(peer);
+		peer_instance = (ao2_ref(peer_instance, -1), NULL);
+		return -1;
+	}
+
+	res = instance->engine->make_compatible(chan, instance, peer, peer_instance);
+
+	ast_channel_unlock(peer);
+
+	peer_instance = (ao2_ref(peer_instance, -1), NULL);
+
+	return res;
+}
+
+int ast_rtp_instance_activate(struct ast_rtp_instance *instance)
+{
+	return instance->engine->activate ? instance->engine->activate(instance) : 0;
+}
+
+void ast_rtp_instance_stun_request(struct ast_rtp_instance *instance, struct sockaddr_in *suggestion, const char *username)
+{
+	if (instance->engine->stun_request) {
+		instance->engine->stun_request(instance, suggestion, username);
+	}
+}
+
+void ast_rtp_instance_set_timeout(struct ast_rtp_instance *instance, int timeout)
+{
+	instance->timeout = timeout;
+}
+
+void ast_rtp_instance_set_hold_timeout(struct ast_rtp_instance *instance, int timeout)
+{
+	instance->holdtimeout = timeout;
+}
+
+int ast_rtp_instance_get_timeout(struct ast_rtp_instance *instance)
+{
+	return instance->timeout;
+}
+
+int ast_rtp_instance_get_hold_timeout(struct ast_rtp_instance *instance)
+{
+	return instance->holdtimeout;
+}