rtp_engine.c

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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: 331886 $")

#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"
#include "asterisk/translate.h"
#include "asterisk/netsock2.h"
#include "asterisk/framehook.h"

struct ast_srtp_res *res_srtp = NULL;
struct ast_srtp_policy_res *res_srtp_policy = NULL;

/*! 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 ast_sockaddr local_address;
   /*! Address that we are sending RTP to */
   struct ast_sockaddr remote_address;
   /*! Alternate address that we are receiving RTP from */
   struct ast_sockaddr alt_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;
   /*! RTP keepalive interval */
   int keepalive;
   /*! Glue currently in use */
   struct ast_rtp_glue *glue;
   /*! Channel associated with the instance */
   struct ast_channel *chan;
   /*! SRTP info associated with the instance */
   struct ast_srtp *srtp;
};

/*! 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_SLINEAR16}, "audio", "L16", 16000},
   {{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_SPEEX16}, "audio", "speex", 16000},
   {{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},
   {{1, AST_FORMAT_G719}, "audio", "G719", 48000},
};

/*!
 * \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},
   [116] = {1, AST_FORMAT_G719},
   [117] = {1, AST_FORMAT_SPEEX16},
   [118] = {1, AST_FORMAT_SLINEAR16}, /* 16 Khz signed linear */
   [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;
   }

   if (instance->srtp) {
      res_srtp->destroy(instance->srtp);
   }

   /* 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, const struct ast_sockaddr *sa,
      void *data)
{
   struct ast_sockaddr address = {{0,}};
   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;
   ast_sockaddr_copy(&instance->local_address, sa);
   ast_sockaddr_copy(&address, sa);

   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)) {
      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,
      const struct ast_sockaddr *address)
{
   ast_sockaddr_copy(&instance->local_address, address);
   return 0;
}

int ast_rtp_instance_set_remote_address(struct ast_rtp_instance *instance,
      const struct ast_sockaddr *address)
{
   ast_sockaddr_copy(&instance->remote_address, address);

   /* moo */

   if (instance->engine->remote_address_set) {
      instance->engine->remote_address_set(instance, &instance->remote_address);
   }

   return 0;
}

int ast_rtp_instance_set_alt_remote_address(struct ast_rtp_instance *instance,
      const struct ast_sockaddr *address)
{
   ast_sockaddr_copy(&instance->alt_remote_address, address);

   /* oink */

   if (instance->engine->alt_remote_address_set) {
      instance->engine->alt_remote_address_set(instance, &instance->alt_remote_address);
   }

   return 0;
}

int ast_rtp_instance_get_and_cmp_local_address(struct ast_rtp_instance *instance,
      struct ast_sockaddr *address)
{
   if (ast_sockaddr_cmp(address, &instance->local_address) != 0) {
      ast_sockaddr_copy(address, &instance->local_address);
      return 1;
   }

   return 0;
}

void ast_rtp_instance_get_local_address(struct ast_rtp_instance *instance,
      struct ast_sockaddr *address)
{
   ast_sockaddr_copy(address, &instance->local_address);
}

int ast_rtp_instance_get_and_cmp_remote_address(struct ast_rtp_instance *instance,
      struct ast_sockaddr *address)
{
   if (ast_sockaddr_cmp(address, &instance->remote_address) != 0) {
      ast_sockaddr_copy(address, &instance->remote_address);
      return 1;
   }

   return 0;
}

void ast_rtp_instance_get_remote_address(struct ast_rtp_instance *instance,
      struct ast_sockaddr *address)
{
   ast_sockaddr_copy(address, &instance->remote_address);
}

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++) {
      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) {
         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, format_t *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 format_t 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) {
         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 format_t 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, format_t 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 format_t capability, const int asterisk_format, enum ast_rtp_options options)
{
   format_t format;
   int found = 0;

   if (!buf) {
      return NULL;
   }

   ast_str_append(&buf, 0, "0x%llx (", (unsigned long long) 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_end_with_duration(struct ast_rtp_instance *instance, char digit, unsigned int duration)
{
   return instance->engine->dtmf_end_with_duration ? instance->engine->dtmf_end_with_duration(instance, digit, duration) : -1;
}

int ast_rtp_instance_dtmf_mode_set(struct ast_rtp_instance *instance, enum ast_rtp_dtmf_mode dtmf_mode)
{
   return (!instance->engine->dtmf_mode_set || instance->engine->dtmf_mode_set(instance, dtmf_mode)) ? -1 : 0;
}

enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get(struct ast_rtp_instance *instance)
{
   return instance->engine->dtmf_mode_get ? instance->engine->dtmf_mode_get(instance) : 0;
}

void ast_rtp_instance_update_source(struct ast_rtp_instance *instance)
{
   if (instance->engine->update_source) {
      instance->engine->update_source(instance);
   }
}

void ast_rtp_instance_change_source(struct ast_rtp_instance *instance)
{
   if (instance->engine->change_source) {
      instance->engine->change_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_framehook_list_is_empty(c0->framehooks) || !ast_framehook_list_is_empty(c1->framehooks))) {
         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.integer == AST_CONTROL_HOLD) ||
             (fr->subclass.integer == AST_CONTROL_UNHOLD) ||
             (fr->subclass.integer == AST_CONTROL_VIDUPDATE) ||
             (fr->subclass.integer == AST_CONTROL_SRCUPDATE) ||
             (fr->subclass.integer == AST_CONTROL_T38_PARAMETERS)) {
            /* If we are going on hold, then break callback mode and P2P bridging */
            if (fr->subclass.integer == 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.integer == 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.integer, fr->data.ptr, fr->datalen);
            ast_frfree(fr);
         } else if (fr->subclass.integer == AST_CONTROL_CONNECTED_LINE) {
            if (ast_channel_connected_line_macro(who, other, fr, other == c0, 1)) {
               ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen);
            }
            ast_frfree(fr);
         } else if (fr->subclass.integer == AST_CONTROL_REDIRECTING) {
            if (ast_channel_redirecting_macro(who, other, fr, other == c0, 1)) {
               ast_indicate_data(other, fr->subclass.integer, 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.integer, 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, format_t codec0, format_t 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, };
   format_t oldcodec0 = codec0, oldcodec1 = codec1;
   struct ast_sockaddr ac1 = {{0,}}, vac1 = {{0,}}, tac1 = {{0,}}, ac0 = {{0,}}, vac0 = {{0,}}, tac0 = {{0,}};
   struct ast_sockaddr 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_framehook_list_is_empty(c0->framehooks) || !ast_framehook_list_is_empty(c1->framehooks))) {
         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 ((ast_sockaddr_cmp(&t1, &ac1)) ||
          (vinstance1 && ast_sockaddr_cmp(&vt1, &vac1)) ||
          (tinstance1 && ast_sockaddr_cmp(&tt1, &tac1)) ||
          (codec1 != oldcodec1)) {
         ast_debug(1, "Oooh, '%s' changed end address to %s (format %s)\n",
              c1->name, ast_sockaddr_stringify(&t1),
              ast_getformatname(codec1));
         ast_debug(1, "Oooh, '%s' changed end vaddress to %s (format %s)\n",
              c1->name, ast_sockaddr_stringify(&vt1),
              ast_getformatname(codec1));
         ast_debug(1, "Oooh, '%s' changed end taddress to %s (format %s)\n",
              c1->name, ast_sockaddr_stringify(&tt1),
              ast_getformatname(codec1));
         ast_debug(1, "Oooh, '%s' was %s/(format %s)\n",
              c1->name, ast_sockaddr_stringify(&ac1),
              ast_getformatname(oldcodec1));
         ast_debug(1, "Oooh, '%s' was %s/(format %s)\n",
              c1->name, ast_sockaddr_stringify(&vac1),
              ast_getformatname(oldcodec1));
         ast_debug(1, "Oooh, '%s' was %s/(format %s)\n",
              c1->name, ast_sockaddr_stringify(&tac1),
              ast_getformatname(oldcodec1));
         if (glue0->update_peer(c0,
                      ast_sockaddr_isnull(&t1)  ? NULL : instance1,
                      ast_sockaddr_isnull(&vt1) ? NULL : vinstance1,
                      ast_sockaddr_isnull(&tt1) ? NULL : tinstance1,
                      codec1, 0)) {
            ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c0->name, c1->name);
         }
         ast_sockaddr_copy(&ac1, &t1);
         ast_sockaddr_copy(&vac1, &vt1);
         ast_sockaddr_copy(&tac1, &tt1);
         oldcodec1 = codec1;
      }
      if ((ast_sockaddr_cmp(&t0, &ac0)) ||
          (vinstance0 && ast_sockaddr_cmp(&vt0, &vac0)) ||
          (tinstance0 && ast_sockaddr_cmp(&tt0, &tac0)) ||
          (codec0 != oldcodec0)) {
         ast_debug(1, "Oooh, '%s' changed end address to %s (format %s)\n",
              c0->name, ast_sockaddr_stringify(&t0),
              ast_getformatname(codec0));
         ast_debug(1, "Oooh, '%s' was %s/(format %s)\n",
              c0->name, ast_sockaddr_stringify(&ac0),
              ast_getformatname(oldcodec0));
         if (glue1->update_peer(c1, t0.len ? instance0 : NULL,
                  vt0.len ? vinstance0 : NULL,
                  tt0.len ? tinstance0 : NULL,
                  codec0, 0)) {
            ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c1->name, c0->name);
         }
         ast_sockaddr_copy(&ac0, &t0);
         ast_sockaddr_copy(&vac0, &vt0);
         ast_sockaddr_copy(&tac0, &tt0);
         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.integer == AST_CONTROL_HOLD) ||
             (fr->subclass.integer == AST_CONTROL_UNHOLD) ||
             (fr->subclass.integer == AST_CONTROL_VIDUPDATE) ||
             (fr->subclass.integer == AST_CONTROL_SRCUPDATE) ||
             (fr->subclass.integer == AST_CONTROL_T38_PARAMETERS)) {
            if (fr->subclass.integer == 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.integer == 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);
            ast_sockaddr_copy(&ac0, &t0);
            ast_rtp_instance_get_remote_address(instance1, &t1);
            ast_sockaddr_copy(&ac1, &t1);
            /* 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.integer, fr->data.ptr, fr->datalen);
            ast_frfree(fr);
         } else if (fr->subclass.integer == AST_CONTROL_CONNECTED_LINE) {
            if (ast_channel_connected_line_macro(who, other, fr, other == c0, 1)) {
               ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen);
            }
            ast_frfree(fr);
         } else if (fr->subclass.integer == AST_CONTROL_REDIRECTING) {
            if (ast_channel_redirecting_macro(who, other, fr, other == c0, 1)) {
               ast_indicate_data(other, fr->subclass.integer, 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.integer, 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 (ast_test_flag(c0, AST_FLAG_ZOMBIE)) {
      ast_debug(1, "Channel '%s' Zombie cleardown from bridge\n", c0->name);
   } else if (c0->tech_pvt != pvt0) {
      ast_debug(1, "Channel c0->'%s' pvt changed, in bridge with c1->'%s'\n", c0->name, c1->name);
   } else if (glue0 != ast_rtp_instance_get_glue(c0->tech->type)) {
      ast_debug(1, "Channel c0->'%s' technology changed, in bridge with c1->'%s'\n", c0->name, c1->name);
   } else 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 (ast_test_flag(c1, AST_FLAG_ZOMBIE)) {
      ast_debug(1, "Channel '%s' Zombie cleardown from bridge\n", c1->name);
   } else if (c1->tech_pvt != pvt1) {
      ast_debug(1, "Channel c1->'%s' pvt changed, in bridge with c0->'%s'\n", c1->name, c0->name);
   } else if (glue1 != ast_rtp_instance_get_glue(c1->tech->type)) {
      ast_debug(1, "Channel c1->'%s' technology changed, in bridge with c0->'%s'\n", c1->name, c0->name);
   } else 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;
   struct ast_sockaddr addr1 = { {0, }, }, addr2 = { {0, }, };
   enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_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;
   enum ast_bridge_result res = AST_BRIDGE_FAILED;
   enum ast_rtp_dtmf_mode dmode;
   format_t 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_debug(1, "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;

   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;

   /* 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 address families differ, force a local bridge */
   ast_rtp_instance_get_remote_address(instance0, &addr1);
   ast_rtp_instance_get_remote_address(instance1, &addr2);

   if (addr1.ss.ss_family != addr2.ss.ss_family ||
      (ast_sockaddr_is_ipv4_mapped(&addr1) != ast_sockaddr_is_ipv4_mapped(&addr2))) {
      audio_glue0_res = AST_RTP_GLUE_RESULT_LOCAL;
      audio_glue1_res = AST_RTP_GLUE_RESULT_LOCAL;
   }

   /* If we need to get DTMF see if we can do it outside of the RTP stream itself */
   dmode = ast_rtp_instance_dtmf_mode_get(instance0);
   if ((flags & AST_BRIDGE_DTMF_CHANNEL_0) && dmode) {
      res = AST_BRIDGE_FAILED_NOWARN;
      goto done;
   }
   dmode = ast_rtp_instance_dtmf_mode_get(instance1);
   if ((flags & AST_BRIDGE_DTMF_CHANNEL_1) && dmode) {
      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 = %s is not codec1 = %s, cannot native bridge in RTP.\n", ast_getformatname(codec0), ast_getformatname(codec1));
      res = AST_BRIDGE_FAILED_NOWARN;
      goto done;
   }

   instance0->glue = glue0;
   instance1->glue = glue1;
   instance0->chan = c0;
   instance1->chan = c1;

   /* 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_verb(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_verb(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);
   }

   instance0->glue = NULL;
   instance1->glue = NULL;
   instance0->chan = NULL;
   instance1->chan = NULL;

   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;
   enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
   format_t 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_debug(1, "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;

   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;

   /* 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) {
      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) {
      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;
   enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
   format_t 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;

   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;

   /* 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 = { 0, };
   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=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f",
          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, format_t 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, format_t 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);
      ao2_ref(peer_instance, -1);
      peer_instance = NULL;
      return -1;
   }

   res = instance->engine->make_compatible(chan, instance, peer, peer_instance);

   ast_channel_unlock(peer);

   ao2_ref(peer_instance, -1);
   peer_instance = NULL;

   return res;
}

format_t ast_rtp_instance_available_formats(struct ast_rtp_instance *instance, format_t to_endpoint, format_t to_asterisk)
{
   format_t formats;

   if (instance->engine->available_formats && (formats = instance->engine->available_formats(instance, to_endpoint, to_asterisk))) {
      return formats;
   }

   return ast_translate_available_formats(to_endpoint, to_asterisk);
}

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 ast_sockaddr *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;
}

void ast_rtp_instance_set_keepalive(struct ast_rtp_instance *instance, int interval)
{
   instance->keepalive = interval;
}

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;
}

int ast_rtp_instance_get_keepalive(struct ast_rtp_instance *instance)
{
   return instance->keepalive;
}

struct ast_rtp_engine *ast_rtp_instance_get_engine(struct ast_rtp_instance *instance)
{
   return instance->engine;
}

struct ast_rtp_glue *ast_rtp_instance_get_active_glue(struct ast_rtp_instance *instance)
{
   return instance->glue;
}

struct ast_channel *ast_rtp_instance_get_chan(struct ast_rtp_instance *instance)
{
   return instance->chan;
}

int ast_rtp_engine_register_srtp(struct ast_srtp_res *srtp_res, struct ast_srtp_policy_res *policy_res)
{
   if (res_srtp || res_srtp_policy) {
      return -1;
   }
   if (!srtp_res || !policy_res) {
      return -1;
   }

   res_srtp = srtp_res;
   res_srtp_policy = policy_res;

   return 0;
}

void ast_rtp_engine_unregister_srtp(void)
{
   res_srtp = NULL;
   res_srtp_policy = NULL;
}

int ast_rtp_engine_srtp_is_registered(void)
{
   return res_srtp && res_srtp_policy;
}

int ast_rtp_instance_add_srtp_policy(struct ast_rtp_instance *instance, struct ast_srtp_policy *policy)
{
   if (!res_srtp) {
      return -1;
   }

   if (!instance->srtp) {
      return res_srtp->create(&instance->srtp, instance, policy);
   } else {
      return res_srtp->add_stream(instance->srtp, policy);
   }
}

struct ast_srtp *ast_rtp_instance_get_srtp(struct ast_rtp_instance *instance)
{
   return instance->srtp;
}

int ast_rtp_instance_sendcng(struct ast_rtp_instance *instance, int level)
{
   if (instance->engine->sendcng) {
      return instance->engine->sendcng(instance, level);
   }

   return -1;
}