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    /*
     * 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"
    
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    /*! 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;
    
    	/*! Alternate address that we are receiving RTP from */
    	struct sockaddr_in alt_remote_address;
    
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    	/*! 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 sockaddr_in address = { 0, };
    
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    	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;
    
    	instance->local_address.sin_family = AF_INET;
    	instance->local_address.sin_addr = sin->sin_addr;
    	instance->remote_address.sin_family = AF_INET;
    
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    	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, &address, data)) {
    
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    		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)
    {
    
    	instance->local_address.sin_addr = address->sin_addr;
    	instance->local_address.sin_port = address->sin_port;
    
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    	return 0;
    }
    
    int ast_rtp_instance_set_remote_address(struct ast_rtp_instance *instance, struct sockaddr_in *address)
    {
    
    	instance->remote_address.sin_addr = address->sin_addr;
    	instance->remote_address.sin_port = address->sin_port;
    
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    	/* moo */
    
    	if (instance->engine->remote_address_set) {
    
    		instance->engine->remote_address_set(instance, &instance->remote_address);
    
    int ast_rtp_instance_set_alt_remote_address(struct ast_rtp_instance *instance, struct sockaddr_in *address)
    {
    	instance->alt_remote_address.sin_addr = address->sin_addr;
    	instance->alt_remote_address.sin_port = address->sin_port;
    
    	/* oink */
    
    	if (instance->engine->alt_remote_address_set) {
    		instance->engine->alt_remote_address_set(instance, &instance->alt_remote_address);
    	}
    
    	return 0;
    }
    
    
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    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));
    
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    		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));
    
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    		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_SRCUPDATE) ||
    			    (fr->subclass == AST_CONTROL_T38_PARAMETERS)) {
    
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    				/* 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) {