CHANNEL(callid): Give dialplan access to the callid.
[asterisk/asterisk.git] / main / rtp_engine.c
index fd448b8..931f89d 100644 (file)
  * \author Joshua Colp <jcolp@digium.com>
  */
 
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+/*** MODULEINFO
+       <support_level>core</support_level>
+***/
+
+/*** DOCUMENTATION
+       <managerEvent language="en_US" name="RTCPSent">
+               <managerEventInstance class="EVENT_FLAG_REPORTING">
+                       <synopsis>Raised when an RTCP packet is sent.</synopsis>
+                       <syntax>
+                               <channel_snapshot/>
+                               <parameter name="SSRC">
+                                       <para>The SSRC identifier for our stream</para>
+                               </parameter>
+                               <parameter name="PT">
+                                       <para>The type of packet for this RTCP report.</para>
+                                       <enumlist>
+                                               <enum name="200(SR)"/>
+                                               <enum name="201(RR)"/>
+                                       </enumlist>
+                               </parameter>
+                               <parameter name="To">
+                                       <para>The address the report is sent to.</para>
+                               </parameter>
+                               <parameter name="ReportCount">
+                                       <para>The number of reports that were sent.</para>
+                                       <para>The report count determines the number of ReportX headers in
+                                       the message. The X for each set of report headers will range from 0 to
+                                       <literal>ReportCount - 1</literal>.</para>
+                               </parameter>
+                               <parameter name="SentNTP" required="false">
+                                       <para>The time the sender generated the report. Only valid when
+                                       PT is <literal>200(SR)</literal>.</para>
+                               </parameter>
+                               <parameter name="SentRTP" required="false">
+                                       <para>The sender's last RTP timestamp. Only valid when PT is
+                                       <literal>200(SR)</literal>.</para>
+                               </parameter>
+                               <parameter name="SentPackets" required="false">
+                                       <para>The number of packets the sender has sent. Only valid when PT
+                                       is <literal>200(SR)</literal>.</para>
+                               </parameter>
+                               <parameter name="SentOctets" required="false">
+                                       <para>The number of bytes the sender has sent. Only valid when PT is
+                                       <literal>200(SR)</literal>.</para>
+                               </parameter>
+                               <parameter name="ReportXSourceSSRC">
+                                       <para>The SSRC for the source of this report block.</para>
+                               </parameter>
+                               <parameter name="ReportXFractionLost">
+                                       <para>The fraction of RTP data packets from <literal>ReportXSourceSSRC</literal>
+                                       lost since the previous SR or RR report was sent.</para>
+                               </parameter>
+                               <parameter name="ReportXCumulativeLost">
+                                       <para>The total number of RTP data packets from <literal>ReportXSourceSSRC</literal>
+                                       lost since the beginning of reception.</para>
+                               </parameter>
+                               <parameter name="ReportXHighestSequence">
+                                       <para>The highest sequence number received in an RTP data packet from
+                                       <literal>ReportXSourceSSRC</literal>.</para>
+                               </parameter>
+                               <parameter name="ReportXSequenceNumberCycles">
+                                       <para>The number of sequence number cycles seen for the RTP data
+                                       received from <literal>ReportXSourceSSRC</literal>.</para>
+                               </parameter>
+                               <parameter name="ReportXIAJitter">
+                                       <para>An estimate of the statistical variance of the RTP data packet
+                                       interarrival time, measured in timestamp units.</para>
+                               </parameter>
+                               <parameter name="ReportXLSR">
+                                       <para>The last SR timestamp received from <literal>ReportXSourceSSRC</literal>.
+                                       If no SR has been received from <literal>ReportXSourceSSRC</literal>,
+                                       then 0.</para>
+                               </parameter>
+                               <parameter name="ReportXDLSR">
+                                       <para>The delay, expressed in units of 1/65536 seconds, between
+                                       receiving the last SR packet from <literal>ReportXSourceSSRC</literal>
+                                       and sending this report.</para>
+                               </parameter>
+                       </syntax>
+                       <see-also>
+                               <ref type="managerEvent">RTCPReceived</ref>
+                       </see-also>
+               </managerEventInstance>
+       </managerEvent>
+       <managerEvent language="en_US" name="RTCPReceived">
+               <managerEventInstance class="EVENT_FLAG_REPORTING">
+                       <synopsis>Raised when an RTCP packet is received.</synopsis>
+                       <syntax>
+                               <channel_snapshot/>
+                               <parameter name="SSRC">
+                                       <para>The SSRC identifier for the remote system</para>
+                               </parameter>
+                               <xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[@name='PT'])" />
+                               <parameter name="From">
+                                       <para>The address the report was received from.</para>
+                               </parameter>
+                               <parameter name="RTT">
+                                       <para>Calculated Round-Trip Time in seconds</para>
+                               </parameter>
+                               <parameter name="ReportCount">
+                                       <para>The number of reports that were received.</para>
+                                       <para>The report count determines the number of ReportX headers in
+                                       the message. The X for each set of report headers will range from 0 to
+                                       <literal>ReportCount - 1</literal>.</para>
+                               </parameter>
+                               <xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[@name='SentNTP'])" />
+                               <xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[@name='SentRTP'])" />
+                               <xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[@name='SentPackets'])" />
+                               <xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[@name='SentOctets'])" />
+                               <xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[contains(@name, 'ReportX')])" />
+                       </syntax>
+                       <see-also>
+                               <ref type="managerEvent">RTCPSent</ref>
+                       </see-also>
+               </managerEventInstance>
+       </managerEvent>
+ ***/
 
-#include <math.h>
+#include "asterisk.h"
 
-#include "asterisk/channel.h"
-#include "asterisk/frame.h"
-#include "asterisk/module.h"
-#include "asterisk/rtp_engine.h"
+#include <math.h>                       /* for sqrt, MAX */
+#include <sched.h>                      /* for sched_yield */
+#include <sys/time.h>                   /* for timeval */
+#include <time.h>                       /* for time_t */
+
+#include "asterisk/_private.h"          /* for ast_rtp_engine_init prototype */
+#include "asterisk/astobj2.h"           /* for ao2_cleanup, ao2_ref, etc */
+#include "asterisk/channel.h"           /* for ast_channel_name, etc */
+#include "asterisk/codec.h"             /* for ast_codec_media_type2str, etc */
+#include "asterisk/format.h"            /* for ast_format_cmp, etc */
+#include "asterisk/format_cache.h"      /* for ast_format_adpcm, etc */
+#include "asterisk/format_cap.h"        /* for ast_format_cap_alloc, etc */
+#include "asterisk/json.h"              /* for ast_json_ref, etc */
+#include "asterisk/linkedlists.h"       /* for ast_rtp_engine::<anonymous>, etc */
+#include "asterisk/lock.h"              /* for ast_rwlock_unlock, etc */
+#include "asterisk/logger.h"            /* for ast_log, ast_debug, etc */
 #include "asterisk/manager.h"
-#include "asterisk/options.h"
-#include "asterisk/astobj2.h"
-#include "asterisk/pbx.h"
+#include "asterisk/module.h"            /* for ast_module_unref, etc */
+#include "asterisk/netsock2.h"          /* for ast_sockaddr_copy, etc */
+#include "asterisk/options.h"           /* for ast_option_rtpptdynamic */
+#include "asterisk/pbx.h"               /* for pbx_builtin_setvar_helper */
+#include "asterisk/res_srtp.h"          /* for ast_srtp_res */
+#include "asterisk/rtp_engine.h"        /* for ast_rtp_codecs, etc */
+#include "asterisk/stasis.h"            /* for stasis_message_data, etc */
+#include "asterisk/stasis_channels.h"   /* for ast_channel_stage_snapshot, etc */
+#include "asterisk/strings.h"           /* for ast_str_append, etc */
+#include "asterisk/time.h"              /* for ast_tvdiff_ms, ast_tvnow */
+#include "asterisk/translate.h"         /* for ast_translate_available_formats */
+#include "asterisk/utils.h"             /* for ast_free, ast_strdup, etc */
+#include "asterisk/vector.h"            /* for AST_VECTOR_GET, etc */
+
+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 {
@@ -47,9 +186,11 @@ struct ast_rtp_instance {
        /*! 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;
+       struct ast_sockaddr local_address;
+       /*! The original source address */
+       struct ast_sockaddr requested_target_address;
        /*! Address that we are sending RTP to */
-       struct sockaddr_in remote_address;
+       struct ast_sockaddr incoming_source_address;
        /*! Instance that we are bridged to if doing remote or local bridging */
        struct ast_rtp_instance *bridged;
        /*! Payload and packetization information */
@@ -58,8 +199,20 @@ struct ast_rtp_instance {
        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;
+       /*! RTP keepalive interval */
+       int keepalive;
+       /*! Glue currently in use */
+       struct ast_rtp_glue *glue;
+       /*! SRTP info associated with the instance */
+       struct ast_srtp *srtp;
+       /*! SRTP info dedicated for RTCP associated with the instance */
+       struct ast_srtp *rtcp_srtp;
+       /*! Channel unique ID */
+       char channel_uniqueid[AST_MAX_UNIQUEID];
+       /*! Time of last packet sent */
+       time_t last_tx;
+       /*! Time of last packet received */
+       time_t last_rx;
 };
 
 /*! List of RTP engines that are currently registered */
@@ -68,49 +221,22 @@ static AST_RWLIST_HEAD_STATIC(engines, ast_rtp_engine);
 /*! List of RTP glues */
 static AST_RWLIST_HEAD_STATIC(glues, ast_rtp_glue);
 
+#define MAX_RTP_MIME_TYPES 128
+
 /*! 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 {
+static struct ast_rtp_mime_type {
+       /*! \brief A mapping object between the Asterisk codec and this RTP payload */
        struct ast_rtp_payload_type payload_type;
-       char *type;
-       char *subtype;
+       /*! \brief The media type */
+       char type[16];
+       /*! \brief The format type */
+       char subtype[64];
+       /*! \brief Expected sample rate of the /c 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},
-};
+} ast_rtp_mime_types[128]; /* This will Likely not need to grow any time soon. */
+static ast_rwlock_t mime_types_lock;
+static int mime_types_len = 0;
 
 /*!
  * \brief Mapping between Asterisk codecs and rtp payload types
@@ -122,43 +248,33 @@ static const struct ast_rtp_mime_type {
  * 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 */
-};
+static struct ast_rtp_payload_type *static_RTP_PT[AST_RTP_MAX_PT];
+static ast_rwlock_t static_RTP_PT_lock;
+
+/*! \brief \ref stasis topic for RTP related messages */
+static struct stasis_topic *rtp_topic;
+
+
+/*!
+ * \internal
+ * \brief Destructor for \c ast_rtp_payload_type
+ */
+static void rtp_payload_type_dtor(void *obj)
+{
+       struct ast_rtp_payload_type *payload = obj;
+
+       ao2_cleanup(payload->format);
+}
+
+struct ast_rtp_payload_type *ast_rtp_engine_alloc_payload_type(void)
+{
+       struct ast_rtp_payload_type *payload;
+
+       payload = ao2_alloc_options(sizeof(*payload), rtp_payload_type_dtor,
+               AO2_ALLOC_OPT_LOCK_NOLOCK);
+
+       return payload;
+}
 
 int ast_rtp_engine_register2(struct ast_rtp_engine *engine, struct ast_module *module)
 {
@@ -263,6 +379,16 @@ static void instance_destructor(void *obj)
                return;
        }
 
+       if (instance->srtp) {
+               res_srtp->destroy(instance->srtp);
+       }
+
+       if (instance->rtcp_srtp) {
+               res_srtp->destroy(instance->rtcp_srtp);
+       }
+
+       ast_rtp_codecs_payloads_destroy(&instance->codecs);
+
        /* Drop our engine reference */
        ast_module_unref(instance->engine->mod);
 
@@ -276,8 +402,11 @@ int ast_rtp_instance_destroy(struct ast_rtp_instance *instance)
        return 0;
 }
 
-struct ast_rtp_instance *ast_rtp_instance_new(const char *engine_name, struct sched_context *sched, struct sockaddr_in *sin, void *data)
+struct ast_rtp_instance *ast_rtp_instance_new(const char *engine_name,
+               struct ast_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;
 
@@ -312,12 +441,18 @@ struct ast_rtp_instance *ast_rtp_instance_new(const char *engine_name, struct sc
                return NULL;
        }
        instance->engine = engine;
-       memcpy(&instance->local_address, sin, sizeof(instance->local_address));
+       ast_sockaddr_copy(&instance->local_address, sa);
+       ast_sockaddr_copy(&address, sa);
+
+       if (ast_rtp_codecs_payloads_initialize(&instance->codecs)) {
+               ao2_ref(instance, -1);
+               return NULL;
+       }
 
        ast_debug(1, "Using engine '%s' for RTP instance '%p'\n", engine->name, instance);
 
        /* And pass it off to the engine to setup */
-       if (instance->engine->new(instance, sched, sin, data)) {
+       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;
@@ -328,6 +463,16 @@ struct ast_rtp_instance *ast_rtp_instance_new(const char *engine_name, struct sc
        return instance;
 }
 
+const char *ast_rtp_instance_get_channel_id(struct ast_rtp_instance *instance)
+{
+       return instance->channel_uniqueid;
+}
+
+void ast_rtp_instance_set_channel_id(struct ast_rtp_instance *instance, const char *uniqueid)
+{
+       ast_copy_string(instance->channel_uniqueid, uniqueid, sizeof(instance->channel_uniqueid));
+}
+
 void ast_rtp_instance_set_data(struct ast_rtp_instance *instance, void *data)
 {
        instance->data = data;
@@ -348,51 +493,75 @@ struct ast_frame *ast_rtp_instance_read(struct ast_rtp_instance *instance, int r
        return instance->engine->read(instance, rtcp);
 }
 
-int ast_rtp_instance_set_local_address(struct ast_rtp_instance *instance, struct sockaddr_in *address)
+int ast_rtp_instance_set_local_address(struct ast_rtp_instance *instance,
+               const struct ast_sockaddr *address)
 {
-       memcpy(&instance->local_address, address, sizeof(instance->local_address));
+       ast_sockaddr_copy(&instance->local_address, address);
        return 0;
 }
 
-int ast_rtp_instance_set_remote_address(struct ast_rtp_instance *instance, struct sockaddr_in *address)
+int ast_rtp_instance_set_incoming_source_address(struct ast_rtp_instance *instance,
+                                                const struct ast_sockaddr *address)
 {
-       if (&instance->remote_address != address) {
-               memcpy(&instance->remote_address, address, sizeof(instance->remote_address));
-       }
+       ast_sockaddr_copy(&instance->incoming_source_address, address);
 
        /* moo */
 
        if (instance->engine->remote_address_set) {
-               instance->engine->remote_address_set(instance, address);
+               instance->engine->remote_address_set(instance, &instance->incoming_source_address);
        }
 
        return 0;
 }
 
-int ast_rtp_instance_get_local_address(struct ast_rtp_instance *instance, struct sockaddr_in *address)
+int ast_rtp_instance_set_requested_target_address(struct ast_rtp_instance *instance,
+                                                 const struct ast_sockaddr *address)
+{
+       ast_sockaddr_copy(&instance->requested_target_address, address);
+
+       return ast_rtp_instance_set_incoming_source_address(instance, address);
+}
+
+int ast_rtp_instance_get_and_cmp_local_address(struct ast_rtp_instance *instance,
+               struct ast_sockaddr *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));
+       if (ast_sockaddr_cmp(address, &instance->local_address) != 0) {
+               ast_sockaddr_copy(address, &instance->local_address);
                return 1;
        }
 
        return 0;
 }
 
-int ast_rtp_instance_get_remote_address(struct ast_rtp_instance *instance, struct sockaddr_in *address)
+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_requested_target_address(struct ast_rtp_instance *instance,
+               struct ast_sockaddr *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));
+       if (ast_sockaddr_cmp(address, &instance->requested_target_address) != 0) {
+               ast_sockaddr_copy(address, &instance->requested_target_address);
                return 1;
        }
 
        return 0;
 }
 
+void ast_rtp_instance_get_incoming_source_address(struct ast_rtp_instance *instance,
+                                                 struct ast_sockaddr *address)
+{
+       ast_sockaddr_copy(address, &instance->incoming_source_address);
+}
+
+void ast_rtp_instance_get_requested_target_address(struct ast_rtp_instance *instance,
+                                                  struct ast_sockaddr *address)
+{
+       ast_sockaddr_copy(address, &instance->requested_target_address);
+}
+
 void ast_rtp_instance_set_extended_prop(struct ast_rtp_instance *instance, int property, void *value)
 {
        if (instance->engine->extended_prop_set) {
@@ -428,66 +597,371 @@ struct ast_rtp_codecs *ast_rtp_instance_get_codecs(struct ast_rtp_instance *inst
        return &instance->codecs;
 }
 
+int ast_rtp_codecs_payloads_initialize(struct ast_rtp_codecs *codecs)
+{
+       int res;
+
+       codecs->framing = 0;
+       ast_rwlock_init(&codecs->codecs_lock);
+       res = AST_VECTOR_INIT(&codecs->payload_mapping_rx, AST_RTP_MAX_PT);
+       res |= AST_VECTOR_INIT(&codecs->payload_mapping_tx, AST_RTP_MAX_PT);
+       if (res) {
+               AST_VECTOR_FREE(&codecs->payload_mapping_rx);
+               AST_VECTOR_FREE(&codecs->payload_mapping_tx);
+       }
+
+       return res;
+}
+
+void ast_rtp_codecs_payloads_destroy(struct ast_rtp_codecs *codecs)
+{
+       int idx;
+       struct ast_rtp_payload_type *type;
+
+       for (idx = 0; idx < AST_VECTOR_SIZE(&codecs->payload_mapping_rx); ++idx) {
+               type = AST_VECTOR_GET(&codecs->payload_mapping_rx, idx);
+               ao2_t_cleanup(type, "destroying ast_rtp_codec rx mapping");
+       }
+       AST_VECTOR_FREE(&codecs->payload_mapping_rx);
+
+       for (idx = 0; idx < AST_VECTOR_SIZE(&codecs->payload_mapping_tx); ++idx) {
+               type = AST_VECTOR_GET(&codecs->payload_mapping_tx, idx);
+               ao2_t_cleanup(type, "destroying ast_rtp_codec tx mapping");
+       }
+       AST_VECTOR_FREE(&codecs->payload_mapping_tx);
+
+       ast_rwlock_destroy(&codecs->codecs_lock);
+}
+
 void ast_rtp_codecs_payloads_clear(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance)
 {
-       int i;
+       ast_rtp_codecs_payloads_destroy(codecs);
+       ast_rtp_codecs_payloads_initialize(codecs);
+
+       if (instance && instance->engine && instance->engine->payload_set) {
+               int i;
+               for (i = 0; i < AST_RTP_MAX_PT; i++) {
+                       instance->engine->payload_set(instance, i, 0, NULL, 0);
+               }
+       }
+}
+
+/*!
+ * \internal
+ * \brief Clear the rx primary mapping flag on all other matching mappings.
+ * \since 14.0.0
+ *
+ * \param codecs Codecs that need rx clearing.
+ * \param to_match Payload type object to compare against.
+ *
+ * \note It is assumed that codecs is write locked before calling.
+ *
+ * \return Nothing
+ */
+static void payload_mapping_rx_clear_primary(struct ast_rtp_codecs *codecs, struct ast_rtp_payload_type *to_match)
+{
+       int idx;
+       struct ast_rtp_payload_type *current;
+       struct ast_rtp_payload_type *new_type;
+       struct timeval now;
+
+       if (!to_match->primary_mapping) {
+               return;
+       }
+
+       now = ast_tvnow();
+       for (idx = 0; idx < AST_VECTOR_SIZE(&codecs->payload_mapping_rx); ++idx) {
+               current = AST_VECTOR_GET(&codecs->payload_mapping_rx, idx);
+
+               if (!current || current == to_match || !current->primary_mapping) {
+                       continue;
+               }
+               if (current->asterisk_format && to_match->asterisk_format) {
+                       if (ast_format_cmp(current->format, to_match->format) == AST_FORMAT_CMP_NOT_EQUAL) {
+                               continue;
+                       }
+               } else if (!current->asterisk_format && !to_match->asterisk_format) {
+                       if (current->rtp_code != to_match->rtp_code) {
+                               continue;
+                       }
+               } else {
+                       continue;
+               }
+
+               /* Replace current with non-primary marked version */
+               new_type = ast_rtp_engine_alloc_payload_type();
+               if (!new_type) {
+                       continue;
+               }
+               *new_type = *current;
+               new_type->primary_mapping = 0;
+               new_type->when_retired = now;
+               ao2_bump(new_type->format);
+               AST_VECTOR_REPLACE(&codecs->payload_mapping_rx, idx, new_type);
+               ao2_ref(current, -1);
+       }
+}
+
+/*!
+ * \internal
+ * \brief Put the new_type into the rx payload type mapping.
+ * \since 14.0.0
+ *
+ * \param codecs Codecs structure to put new_type into
+ * \param payload type position to replace.
+ * \param new_type RTP payload mapping object to store.
+ *
+ * \note It is assumed that codecs is write locked before calling.
+ *
+ * \return Nothing
+ */
+static void rtp_codecs_payload_replace_rx(struct ast_rtp_codecs *codecs, int payload, struct ast_rtp_payload_type *new_type)
+{
+       ao2_ref(new_type, +1);
+       if (payload < AST_VECTOR_SIZE(&codecs->payload_mapping_rx)) {
+               ao2_t_cleanup(AST_VECTOR_GET(&codecs->payload_mapping_rx, payload),
+                       "cleaning up rx mapping vector element about to be replaced");
+       }
+       AST_VECTOR_REPLACE(&codecs->payload_mapping_rx, payload, new_type);
+
+       payload_mapping_rx_clear_primary(codecs, new_type);
+}
+
+/*!
+ * \internal
+ * \brief Copy the rx payload type mapping to the destination.
+ * \since 14.0.0
+ *
+ * \param src The source codecs structure
+ * \param dest The destination codecs structure that the values from src will be copied to
+ * \param instance Optionally the instance that the dst codecs structure belongs to
+ *
+ * \note It is assumed that src is at least read locked before calling.
+ * \note It is assumed that dest is write locked before calling.
+ *
+ * \return Nothing
+ */
+static void rtp_codecs_payloads_copy_rx(struct ast_rtp_codecs *src, struct ast_rtp_codecs *dest, struct ast_rtp_instance *instance)
+{
+       int idx;
+       struct ast_rtp_payload_type *type;
+
+       for (idx = 0; idx < AST_VECTOR_SIZE(&src->payload_mapping_rx); ++idx) {
+               type = AST_VECTOR_GET(&src->payload_mapping_rx, idx);
+               if (!type) {
+                       continue;
+               }
+
+               ast_debug(2, "Copying rx payload mapping %d (%p) from %p to %p\n",
+                       idx, type, src, dest);
+               rtp_codecs_payload_replace_rx(dest, idx, type);
 
-       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);
+                       instance->engine->payload_set(instance, idx, type->asterisk_format, type->format, type->rtp_code);
                }
        }
 }
 
-void ast_rtp_codecs_payloads_default(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance)
+/*!
+ * \internal
+ * \brief Determine if a type of payload is already present in mappings.
+ * \since 14.0.0
+ *
+ * \param codecs Codecs to be checked for mappings.
+ * \param to_match Payload type object to compare against.
+ *
+ * \note It is assumed that codecs is write locked before calling.
+ *
+ * \retval 0 not found
+ * \retval 1 found
+ */
+static int payload_mapping_tx_is_present(const struct ast_rtp_codecs *codecs, const struct ast_rtp_payload_type *to_match)
 {
-       int i;
+       int idx;
+       struct ast_rtp_payload_type *current;
 
-       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);
+       for (idx = 0; idx < AST_VECTOR_SIZE(&codecs->payload_mapping_tx); ++idx) {
+               current = AST_VECTOR_GET(&codecs->payload_mapping_tx, idx);
+
+               if (!current) {
+                       continue;
+               }
+               if (current == to_match) {
+                       /* The exact object is already in the mapping. */
+                       return 1;
+               }
+               if (current->asterisk_format && to_match->asterisk_format) {
+                       if (ast_format_get_codec_id(current->format) != ast_format_get_codec_id(to_match->format)) {
+                               continue;
+                       } else if (current->payload == to_match->payload) {
+                               return 0;
                        }
+               } else if (!current->asterisk_format && !to_match->asterisk_format) {
+                       if (current->rtp_code != to_match->rtp_code) {
+                               continue;
+                       }
+               } else {
+                       continue;
+               }
+
+               return 1;
+       }
+
+       return 0;
+}
+
+/*!
+ * \internal
+ * \brief Copy the tx payload type mapping to the destination.
+ * \since 14.0.0
+ *
+ * \param src The source codecs structure
+ * \param dest The destination codecs structure that the values from src will be copied to
+ * \param instance Optionally the instance that the dst codecs structure belongs to
+ *
+ * \note It is assumed that src is at least read locked before calling.
+ * \note It is assumed that dest is write locked before calling.
+ *
+ * \return Nothing
+ */
+static void rtp_codecs_payloads_copy_tx(struct ast_rtp_codecs *src, struct ast_rtp_codecs *dest, struct ast_rtp_instance *instance)
+{
+       int idx;
+       struct ast_rtp_payload_type *type;
+
+       for (idx = 0; idx < AST_VECTOR_SIZE(&src->payload_mapping_tx); ++idx) {
+               type = AST_VECTOR_GET(&src->payload_mapping_tx, idx);
+               if (!type) {
+                       continue;
+               }
+
+               ast_debug(2, "Copying tx payload mapping %d (%p) from %p to %p\n",
+                       idx, type, src, dest);
+               ao2_ref(type, +1);
+               if (idx < AST_VECTOR_SIZE(&dest->payload_mapping_tx)) {
+                       ao2_t_cleanup(AST_VECTOR_GET(&dest->payload_mapping_tx, idx),
+                               "cleaning up tx mapping vector element about to be replaced");
+               }
+               AST_VECTOR_REPLACE(&dest->payload_mapping_tx, idx, type);
+
+               if (instance && instance->engine && instance->engine->payload_set) {
+                       instance->engine->payload_set(instance, idx, type->asterisk_format, type->format, type->rtp_code);
                }
        }
 }
 
 void ast_rtp_codecs_payloads_copy(struct ast_rtp_codecs *src, struct ast_rtp_codecs *dest, struct ast_rtp_instance *instance)
 {
-       int i;
+       int idx;
+       struct ast_rtp_payload_type *type;
 
-       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);
-                       }
+       ast_rwlock_wrlock(&dest->codecs_lock);
+
+       /* Deadlock avoidance because of held write lock. */
+       while (ast_rwlock_tryrdlock(&src->codecs_lock)) {
+               ast_rwlock_unlock(&dest->codecs_lock);
+               sched_yield();
+               ast_rwlock_wrlock(&dest->codecs_lock);
+       }
+
+       /*
+        * This represents a completely new mapping of what the remote party is
+        * expecting for payloads, so we clear out the entire tx payload mapping
+        * vector and replace it.
+        */
+       for (idx = 0; idx < AST_VECTOR_SIZE(&dest->payload_mapping_tx); ++idx) {
+               type = AST_VECTOR_GET(&dest->payload_mapping_tx, idx);
+               ao2_t_cleanup(type, "destroying ast_rtp_codec tx mapping");
+               AST_VECTOR_REPLACE(&dest->payload_mapping_tx, idx, NULL);
+       }
+
+       rtp_codecs_payloads_copy_rx(src, dest, instance);
+       rtp_codecs_payloads_copy_tx(src, dest, instance);
+       dest->framing = src->framing;
+
+       ast_rwlock_unlock(&src->codecs_lock);
+       ast_rwlock_unlock(&dest->codecs_lock);
+}
+
+void ast_rtp_codecs_payloads_xover(struct ast_rtp_codecs *src, struct ast_rtp_codecs *dest, struct ast_rtp_instance *instance)
+{
+       int idx;
+       struct ast_rtp_payload_type *type;
+
+       ast_rwlock_wrlock(&dest->codecs_lock);
+       if (src != dest) {
+               /* Deadlock avoidance because of held write lock. */
+               while (ast_rwlock_tryrdlock(&src->codecs_lock)) {
+                       ast_rwlock_unlock(&dest->codecs_lock);
+                       sched_yield();
+                       ast_rwlock_wrlock(&dest->codecs_lock);
+               }
+       }
+
+       /* Crossover copy payload type tx mapping to rx mapping. */
+       for (idx = 0; idx < AST_VECTOR_SIZE(&src->payload_mapping_tx); ++idx) {
+               type = AST_VECTOR_GET(&src->payload_mapping_tx, idx);
+               if (!type) {
+                       continue;
+               }
+
+               /* All tx mapping elements should have the primary flag set. */
+               ast_assert(type->primary_mapping);
+
+               ast_debug(2, "Crossover copying tx to rx payload mapping %d (%p) from %p to %p\n",
+                       idx, type, src, dest);
+               rtp_codecs_payload_replace_rx(dest, idx, type);
+
+               if (instance && instance->engine && instance->engine->payload_set) {
+                       instance->engine->payload_set(instance, idx, type->asterisk_format, type->format, type->rtp_code);
                }
        }
+
+       dest->framing = src->framing;
+
+       if (src != dest) {
+               ast_rwlock_unlock(&src->codecs_lock);
+       }
+       ast_rwlock_unlock(&dest->codecs_lock);
 }
 
 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) {
+       struct ast_rtp_payload_type *new_type;
+
+       if (payload < 0 || payload >= AST_RTP_MAX_PT) {
+               return;
+       }
+
+       ast_rwlock_rdlock(&static_RTP_PT_lock);
+       new_type = ao2_bump(static_RTP_PT[payload]);
+       ast_rwlock_unlock(&static_RTP_PT_lock);
+       if (!new_type) {
+               ast_debug(1, "Don't have a default tx payload type %d format for m type on %p\n",
+                       payload, codecs);
                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 tx payload type %d based on m type on %p\n",
+               payload, codecs);
 
-       ast_debug(1, "Setting payload %d based on m type on %p\n", payload, codecs);
+       ast_rwlock_wrlock(&codecs->codecs_lock);
 
-       if (instance && instance->engine && instance->engine->payload_set) {
-               instance->engine->payload_set(instance, payload, codecs->payloads[payload].asterisk_format, codecs->payloads[payload].code);
+       if (!payload_mapping_tx_is_present(codecs, new_type)) {
+               if (payload < AST_VECTOR_SIZE(&codecs->payload_mapping_tx)) {
+                       ao2_t_cleanup(AST_VECTOR_GET(&codecs->payload_mapping_tx, payload),
+                               "cleaning up replaced tx payload type");
+               }
+               AST_VECTOR_REPLACE(&codecs->payload_mapping_tx, payload, new_type);
+
+               if (instance && instance->engine && instance->engine->payload_set) {
+                       instance->engine->payload_set(instance, payload, new_type->asterisk_format, new_type->format, new_type->rtp_code);
+               }
+       } else {
+               ao2_ref(new_type, -1);
        }
+
+       ast_rwlock_unlock(&codecs->codecs_lock);
 }
 
 int ast_rtp_codecs_payloads_set_rtpmap_type_rate(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int pt,
@@ -495,14 +969,19 @@ int ast_rtp_codecs_payloads_set_rtpmap_type_rate(struct ast_rtp_codecs *codecs,
                                 enum ast_rtp_options options,
                                 unsigned int sample_rate)
 {
-       unsigned int i;
+       unsigned int idx;
        int found = 0;
 
-       if (pt < 0 || pt > AST_RTP_MAX_PT)
+       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];
+       ast_rwlock_rdlock(&mime_types_lock);
+       ast_rwlock_wrlock(&codecs->codecs_lock);
+
+       for (idx = 0; idx < mime_types_len; ++idx) {
+               const struct ast_rtp_mime_type *t = &ast_rtp_mime_types[idx];
+               struct ast_rtp_payload_type *new_type;
 
                if (strcasecmp(mimesubtype, t->subtype)) {
                        continue;
@@ -513,29 +992,57 @@ int ast_rtp_codecs_payloads_set_rtpmap_type_rate(struct ast_rtp_codecs *codecs,
                }
 
                /* 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 */
+                * 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;
+               new_type = ast_rtp_engine_alloc_payload_type();
+               if (!new_type) {
+                       continue;
+               }
+
+               new_type->asterisk_format = t->payload_type.asterisk_format;
+               new_type->rtp_code = t->payload_type.rtp_code;
+               new_type->payload = pt;
+               new_type->primary_mapping = 1;
+               if (t->payload_type.asterisk_format
+                       && ast_format_cmp(t->payload_type.format, ast_format_g726) == AST_FORMAT_CMP_EQUAL
+                       && (options & AST_RTP_OPT_G726_NONSTANDARD)) {
+                       new_type->format = ast_format_g726_aal2;
+               } else {
+                       new_type->format = t->payload_type.format;
                }
 
-               if (instance && instance->engine && instance->engine->payload_set) {
-                       instance->engine->payload_set(instance, pt, codecs->payloads[i].asterisk_format, codecs->payloads[i].code);
+               if (new_type->format) {
+                       /* SDP parsing automatically increases the reference count */
+                       new_type->format = ast_format_parse_sdp_fmtp(new_type->format, "");
+               }
+
+               if (!payload_mapping_tx_is_present(codecs, new_type)) {
+                       if (pt < AST_VECTOR_SIZE(&codecs->payload_mapping_tx)) {
+                               ao2_t_cleanup(AST_VECTOR_GET(&codecs->payload_mapping_tx, pt),
+                                       "cleaning up replaced tx payload type");
+                       }
+                       AST_VECTOR_REPLACE(&codecs->payload_mapping_tx, pt, new_type);
+
+                       if (instance && instance->engine && instance->engine->payload_set) {
+                               instance->engine->payload_set(instance, pt, new_type->asterisk_format, new_type->format, new_type->rtp_code);
+                       }
+               } else {
+                       ao2_ref(new_type, -1);
                }
 
                break;
        }
 
+       ast_rwlock_unlock(&codecs->codecs_lock);
+       ast_rwlock_unlock(&mime_types_lock);
+
        return (found ? 0 : -2);
 }
 
@@ -546,579 +1053,605 @@ int ast_rtp_codecs_payloads_set_rtpmap_type(struct ast_rtp_codecs *codecs, struc
 
 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) {
+       struct ast_rtp_payload_type *type;
+
+       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;
+       ast_rwlock_wrlock(&codecs->codecs_lock);
+
+       if (payload < AST_VECTOR_SIZE(&codecs->payload_mapping_tx)) {
+               type = AST_VECTOR_GET(&codecs->payload_mapping_tx, payload);
+               ao2_cleanup(type);
+               AST_VECTOR_REPLACE(&codecs->payload_mapping_tx, payload, NULL);
+       }
 
        if (instance && instance->engine && instance->engine->payload_set) {
-               instance->engine->payload_set(instance, payload, 0, 0);
+               instance->engine->payload_set(instance, payload, 0, NULL, 0);
        }
+
+       ast_rwlock_unlock(&codecs->codecs_lock);
 }
 
-struct ast_rtp_payload_type ast_rtp_codecs_payload_lookup(struct ast_rtp_codecs *codecs, int payload)
+struct ast_rtp_payload_type *ast_rtp_codecs_get_payload(struct ast_rtp_codecs *codecs, int payload)
 {
-       struct ast_rtp_payload_type result = { .asterisk_format = 0, };
+       struct ast_rtp_payload_type *type = NULL;
 
-       if (payload < 0 || payload > AST_RTP_MAX_PT) {
-               return result;
+       if (payload < 0 || payload >= AST_RTP_MAX_PT) {
+               return NULL;
        }
 
-       result.asterisk_format = codecs->payloads[payload].asterisk_format;
-       result.code = codecs->payloads[payload].code;
+       ast_rwlock_rdlock(&codecs->codecs_lock);
+       if (payload < AST_VECTOR_SIZE(&codecs->payload_mapping_rx)) {
+               type = AST_VECTOR_GET(&codecs->payload_mapping_rx, payload);
+               ao2_bump(type);
+       }
+       ast_rwlock_unlock(&codecs->codecs_lock);
 
-       if (!result.code) {
-               result = static_RTP_PT[payload];
+       if (!type) {
+               ast_rwlock_rdlock(&static_RTP_PT_lock);
+               type = ao2_bump(static_RTP_PT[payload]);
+               ast_rwlock_unlock(&static_RTP_PT_lock);
        }
 
-       return result;
+       return type;
 }
 
-void ast_rtp_codecs_payload_formats(struct ast_rtp_codecs *codecs, int *astformats, int *nonastformats)
+int ast_rtp_codecs_payload_replace_format(struct ast_rtp_codecs *codecs, int payload, struct ast_format *format)
 {
-       int i;
+       struct ast_rtp_payload_type *type;
 
-       *astformats = *nonastformats = 0;
+       if (payload < 0 || payload >= AST_RTP_MAX_PT || !format) {
+               return -1;
+       }
 
-       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;
+       type = ast_rtp_engine_alloc_payload_type();
+       if (!type) {
+               return -1;
+       }
+       ao2_ref(format, +1);
+       type->format = format;
+       type->asterisk_format = 1;
+       type->payload = payload;
+       type->primary_mapping = 1;
+
+       ast_rwlock_wrlock(&codecs->codecs_lock);
+       if (!payload_mapping_tx_is_present(codecs, type)) {
+               if (payload < AST_VECTOR_SIZE(&codecs->payload_mapping_tx)) {
+                       ao2_cleanup(AST_VECTOR_GET(&codecs->payload_mapping_tx, payload));
                }
+               AST_VECTOR_REPLACE(&codecs->payload_mapping_tx, payload, type);
+       } else {
+               ao2_ref(type, -1);
        }
+       ast_rwlock_unlock(&codecs->codecs_lock);
+
+       return 0;
 }
 
-int ast_rtp_codecs_payload_code(struct ast_rtp_codecs *codecs, const int asterisk_format, const int code)
+struct ast_format *ast_rtp_codecs_get_payload_format(struct ast_rtp_codecs *codecs, int payload)
 {
-       int i;
+       struct ast_rtp_payload_type *type;
+       struct ast_format *format = NULL;
 
-       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;
-               }
+       if (payload < 0 || payload >= AST_RTP_MAX_PT) {
+               return NULL;
        }
 
-       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;
+       ast_rwlock_rdlock(&codecs->codecs_lock);
+       if (payload < AST_VECTOR_SIZE(&codecs->payload_mapping_tx)) {
+               type = AST_VECTOR_GET(&codecs->payload_mapping_tx, payload);
+               if (type && type->asterisk_format) {
+                       format = ao2_bump(type->format);
                }
        }
+       ast_rwlock_unlock(&codecs->codecs_lock);
 
-       return -1;
+       return format;
 }
 
-const char *ast_rtp_lookup_mime_subtype2(const int asterisk_format, const int code, enum ast_rtp_options options)
+void ast_rtp_codecs_set_framing(struct ast_rtp_codecs *codecs, unsigned int framing)
 {
-       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;
-                       }
-               }
+       if (!framing) {
+               return;
        }
 
-       return "";
+       ast_rwlock_wrlock(&codecs->codecs_lock);
+       codecs->framing = framing;
+       ast_rwlock_unlock(&codecs->codecs_lock);
 }
 
-unsigned int ast_rtp_lookup_sample_rate2(int asterisk_format, int code)
+unsigned int ast_rtp_codecs_get_framing(struct ast_rtp_codecs *codecs)
 {
-       unsigned int i;
+       unsigned int framing;
 
-       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;
-               }
-       }
+       ast_rwlock_rdlock(&codecs->codecs_lock);
+       framing = codecs->framing;
+       ast_rwlock_unlock(&codecs->codecs_lock);
 
-       return 0;
+       return framing;
 }
 
-char *ast_rtp_lookup_mime_multiple2(struct ast_str *buf, const int capability, const int asterisk_format, enum ast_rtp_options options)
+void ast_rtp_codecs_payload_formats(struct ast_rtp_codecs *codecs, struct ast_format_cap *astformats, int *nonastformats)
 {
-       int format, found = 0;
+       int idx;
 
-       if (!buf) {
-               return NULL;
-       }
+       ast_format_cap_remove_by_type(astformats, AST_MEDIA_TYPE_UNKNOWN);
+       *nonastformats = 0;
 
-       ast_str_append(&buf, 0, "0x%x (", capability);
+       ast_rwlock_rdlock(&codecs->codecs_lock);
 
-       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;
+       for (idx = 0; idx < AST_VECTOR_SIZE(&codecs->payload_mapping_tx); ++idx) {
+               struct ast_rtp_payload_type *type;
+
+               type = AST_VECTOR_GET(&codecs->payload_mapping_tx, idx);
+               if (!type) {
+                       continue;
                }
-       }
 
-       ast_str_append(&buf, 0, "%s", found ? ")" : "nothing)");
+               if (type->asterisk_format) {
+                       ast_format_cap_append(astformats, type->format, 0);
+               } else {
+                       *nonastformats |= type->rtp_code;
+               }
+       }
+       if (codecs->framing) {
+               ast_format_cap_set_framing(astformats, codecs->framing);
+       }
 
-       return ast_str_buffer(buf);
+       ast_rwlock_unlock(&codecs->codecs_lock);
 }
 
-void ast_rtp_codecs_packetization_set(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, struct ast_codec_pref *prefs)
+/*!
+ * \internal
+ * \brief Find the static payload type mapping for the format.
+ * \since 14.0.0
+ *
+ * \param asterisk_format Non-zero if the given Asterisk format is present
+ * \param format Asterisk format to look for
+ * \param code The non-Asterisk format code to look for
+ *
+ * \note It is assumed that static_RTP_PT_lock is at least read locked before calling.
+ *
+ * \retval Numerical payload type
+ * \retval -1 if not found.
+ */
+static int find_static_payload_type(int asterisk_format, const struct ast_format *format, int code)
 {
-       codecs->pref = *prefs;
+       int idx;
+       int payload = -1;
 
-       if (instance && instance->engine->packetization_set) {
-               instance->engine->packetization_set(instance, &instance->codecs.pref);
-       }
-}
+       if (!asterisk_format) {
+               for (idx = 0; idx < AST_RTP_MAX_PT; ++idx) {
+                       if (static_RTP_PT[idx]
+                               && !static_RTP_PT[idx]->asterisk_format
+                               && static_RTP_PT[idx]->rtp_code == code) {
+                               payload = idx;
+                               break;
+                       }
+               }
+       } else if (format) {
+               for (idx = 0; idx < AST_RTP_MAX_PT; ++idx) {
+                       if (static_RTP_PT[idx]
+                               && static_RTP_PT[idx]->asterisk_format
+                               && ast_format_cmp(format, static_RTP_PT[idx]->format)
+                                       != AST_FORMAT_CMP_NOT_EQUAL) {
+                               payload = idx;
+                               break;
+                       }
+               }
+       }
 
-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;
+       return payload;
 }
 
-int ast_rtp_instance_dtmf_end(struct ast_rtp_instance *instance, char digit)
+/*!
+ * \internal
+ * \brief Find the first unused dynamic rx payload type.
+ * \since 14.0.0
+ *
+ * \param codecs Codecs structure to look in
+ *
+ * \note It is assumed that codecs is at least read locked before calling.
+ *
+ * \retval Numerical payload type
+ * \retval -1 if not found.
+ */
+static int rtp_codecs_find_empty_dynamic_rx(struct ast_rtp_codecs *codecs)
 {
-       return instance->engine->dtmf_end ? instance->engine->dtmf_end(instance, digit) : -1;
+       struct ast_rtp_payload_type *type;
+       int idx;
+       int payload = -1;
+
+       idx = AST_RTP_PT_FIRST_DYNAMIC;
+       for (; idx < AST_VECTOR_SIZE(&codecs->payload_mapping_rx); ++idx) {
+               type = AST_VECTOR_GET(&codecs->payload_mapping_rx, idx);
+               if (!type) {
+                       payload = idx;
+                       break;
+               }
+       }
+       return payload;
 }
 
-int ast_rtp_instance_dtmf_mode_set(struct ast_rtp_instance *instance, enum ast_rtp_dtmf_mode dtmf_mode)
+/*!
+ * \internal
+ * \brief Find the oldest non-primary dynamic rx payload type.
+ * \since 14.0.0
+ *
+ * \param codecs Codecs structure to look in
+ *
+ * \note It is assumed that codecs is at least read locked before calling.
+ *
+ * \retval Numerical payload type
+ * \retval -1 if not found.
+ */
+static int rtp_codecs_find_non_primary_dynamic_rx(struct ast_rtp_codecs *codecs)
+{
+       struct ast_rtp_payload_type *type;
+       struct timeval oldest;
+       int idx;
+       int payload = -1;
+
+       idx = AST_RTP_PT_FIRST_DYNAMIC;
+       for (; idx < AST_VECTOR_SIZE(&codecs->payload_mapping_rx); ++idx) {
+               type = AST_VECTOR_GET(&codecs->payload_mapping_rx, idx);
+               if (type
+                       && !type->primary_mapping
+                       && (payload == -1
+                               || ast_tvdiff_ms(type->when_retired, oldest) < 0)) {
+                       oldest = type->when_retired;
+                       payload = idx;
+               }
+       }
+       return payload;
+}
+
+/*!
+ * \internal
+ * \brief Assign a payload type for the rx mapping.
+ * \since 14.0.0
+ *
+ * \param codecs Codecs structure to look in
+ * \param asterisk_format Non-zero if the given Asterisk format is present
+ * \param format Asterisk format to look for
+ * \param code The format to look for
+ *
+ * \note It is assumed that static_RTP_PT_lock is at least read locked before calling.
+ *
+ * \retval Numerical payload type
+ * \retval -1 if could not assign.
+ */
+static int rtp_codecs_assign_payload_code_rx(struct ast_rtp_codecs *codecs, int asterisk_format, struct ast_format *format, int code)
 {
-       if (!instance->engine->dtmf_mode_set || instance->engine->dtmf_mode_set(instance, dtmf_mode)) {
+       int payload;
+       struct ast_rtp_payload_type *new_type;
+
+       payload = find_static_payload_type(asterisk_format, format, code);
+       if (payload < 0) {
+               return payload;
+       }
+
+       new_type = ast_rtp_engine_alloc_payload_type();
+       if (!new_type) {
                return -1;
        }
+       new_type->format = ao2_bump(format);
+       new_type->asterisk_format = asterisk_format;
+       new_type->rtp_code = code;
+       new_type->payload = payload;
+       new_type->primary_mapping = 1;
+
+       ast_rwlock_wrlock(&codecs->codecs_lock);
+       if (payload < AST_RTP_PT_FIRST_DYNAMIC
+               || AST_VECTOR_SIZE(&codecs->payload_mapping_rx) <= payload
+               || !AST_VECTOR_GET(&codecs->payload_mapping_rx, payload)) {
+               /*
+                * The payload type is a static assignment
+                * or our default dynamic position is available.
+                */
+               rtp_codecs_payload_replace_rx(codecs, payload, new_type);
+       } else if (-1 < (payload = rtp_codecs_find_empty_dynamic_rx(codecs))
+               || -1 < (payload = rtp_codecs_find_non_primary_dynamic_rx(codecs))) {
+               /*
+                * We found the first available empty dynamic position
+                * or we found a mapping that should no longer be
+                * actively used.
+                */
+               new_type->payload = payload;
+               rtp_codecs_payload_replace_rx(codecs, payload, new_type);
+       } else {
+               /*
+                * There are no empty or non-primary dynamic positions
+                * left.  Sadness.
+                *
+                * I don't think this is really possible.
+                */
+               ast_log(LOG_WARNING, "No dynamic RTP payload type values available!\n");
+       }
+       ast_rwlock_unlock(&codecs->codecs_lock);
 
-       instance->dtmf_mode = dtmf_mode;
+       ao2_ref(new_type, -1);
 
-       return 0;
+       return payload;
 }
 
-enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get(struct ast_rtp_instance *instance)
+int ast_rtp_codecs_payload_code(struct ast_rtp_codecs *codecs, int asterisk_format, struct ast_format *format, int code)
 {
-       return instance->dtmf_mode;
-}
+       struct ast_rtp_payload_type *type;
+       int idx;
+       int payload = -1;
 
-void ast_rtp_instance_new_source(struct ast_rtp_instance *instance)
-{
-       if (instance->engine->new_source) {
-               instance->engine->new_source(instance);
-       }
-}
+       ast_rwlock_rdlock(&static_RTP_PT_lock);
+       if (!asterisk_format) {
+               ast_rwlock_rdlock(&codecs->codecs_lock);
+               for (idx = 0; idx < AST_VECTOR_SIZE(&codecs->payload_mapping_rx); ++idx) {
+                       type = AST_VECTOR_GET(&codecs->payload_mapping_rx, idx);
+                       if (!type) {
+                               continue;
+                       }
 
-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;
-}
+                       if (!type->asterisk_format
+                               && type->primary_mapping
+                               && type->rtp_code == code) {
+                               payload = idx;
+                               break;
+                       }
+               }
+               ast_rwlock_unlock(&codecs->codecs_lock);
+       } else if (format) {
+               ast_rwlock_rdlock(&codecs->codecs_lock);
+               for (idx = 0; idx < AST_VECTOR_SIZE(&codecs->payload_mapping_rx); ++idx) {
+                       type = AST_VECTOR_GET(&codecs->payload_mapping_rx, idx);
+                       if (!type) {
+                               continue;
+                       }
 
-void ast_rtp_instance_stop(struct ast_rtp_instance *instance)
-{
-       if (instance->engine->stop) {
-               instance->engine->stop(instance);
+                       if (type->asterisk_format
+                               && type->primary_mapping
+                               && ast_format_cmp(format, type->format) == AST_FORMAT_CMP_EQUAL) {
+                               payload = idx;
+                               break;
+                       }
+               }
+               ast_rwlock_unlock(&codecs->codecs_lock);
        }
-}
 
-int ast_rtp_instance_fd(struct ast_rtp_instance *instance, int rtcp)
-{
-       return instance->engine->fd ? instance->engine->fd(instance, rtcp) : -1;
+       if (payload < 0) {
+               payload = rtp_codecs_assign_payload_code_rx(codecs, asterisk_format, format,
+                       code);
+       }
+       ast_rwlock_unlock(&static_RTP_PT_lock);
+
+       return payload;
 }
 
-struct ast_rtp_glue *ast_rtp_instance_get_glue(const char *type)
+int ast_rtp_codecs_payload_code_tx(struct ast_rtp_codecs *codecs, int asterisk_format, const struct ast_format *format, int code)
 {
-       struct ast_rtp_glue *glue = NULL;
+       struct ast_rtp_payload_type *type;
+       int idx;
+       int payload = -1;
 
-       AST_RWLIST_RDLOCK(&glues);
+       if (!asterisk_format) {
+               ast_rwlock_rdlock(&codecs->codecs_lock);
+               for (idx = 0; idx < AST_VECTOR_SIZE(&codecs->payload_mapping_tx); ++idx) {
+                       type = AST_VECTOR_GET(&codecs->payload_mapping_tx, idx);
+                       if (!type) {
+                               continue;
+                       }
 
-       AST_RWLIST_TRAVERSE(&glues, glue, entry) {
-               if (!strcasecmp(glue->type, type)) {
-                       break;
+                       if (!type->asterisk_format
+                               && type->rtp_code == code) {
+                               payload = idx;
+                               break;
+                       }
+               }
+               ast_rwlock_unlock(&codecs->codecs_lock);
+       } else if (format) {
+               ast_rwlock_rdlock(&codecs->codecs_lock);
+               for (idx = 0; idx < AST_VECTOR_SIZE(&codecs->payload_mapping_tx); ++idx) {
+                       type = AST_VECTOR_GET(&codecs->payload_mapping_tx, idx);
+                       if (!type) {
+                               continue;
+                       }
+
+                       if (type->asterisk_format
+                               && ast_format_cmp(format, type->format) == AST_FORMAT_CMP_EQUAL) {
+                               payload = idx;
+                               break;
+                       }
                }
+               ast_rwlock_unlock(&codecs->codecs_lock);
        }
 
-       AST_RWLIST_UNLOCK(&glues);
+       if (payload < 0) {
+               ast_rwlock_rdlock(&static_RTP_PT_lock);
+               payload = find_static_payload_type(asterisk_format, format, code);
+               ast_rwlock_unlock(&static_RTP_PT_lock);
+       }
 
-       return glue;
+       return payload;
 }
 
-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)
+int ast_rtp_codecs_find_payload_code(struct ast_rtp_codecs *codecs, int payload)
 {
-       enum ast_bridge_result res = AST_BRIDGE_FAILED;
-       struct ast_channel *who = NULL, *other = NULL, *cs[3] = { NULL, };
-       struct ast_frame *fr = NULL;
+       struct ast_rtp_payload_type *type;
+       int res = -1;
 
-       /* 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_rwlock_rdlock(&codecs->codecs_lock);
+       if (payload < AST_VECTOR_SIZE(&codecs->payload_mapping_tx)) {
+               type = AST_VECTOR_GET(&codecs->payload_mapping_tx, payload);
+               if (type) {
+                       res = payload;
                }
-               ast_channel_unlock(c0);
-               ast_channel_unlock(c1);
-               return AST_BRIDGE_FAILED_NOWARN;
        }
+       ast_rwlock_unlock(&codecs->codecs_lock);
 
-       ast_channel_unlock(c0);
-       ast_channel_unlock(c1);
-
-       instance0->bridged = instance1;
-       instance1->bridged = instance0;
-
-       ast_poll_channel_add(c0, c1);
+       return res;
+}
 
-       /* 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)) {
+const char *ast_rtp_lookup_mime_subtype2(const int asterisk_format,
+       const struct ast_format *format, int code, enum ast_rtp_options options)
+{
+       int i;
+       const char *res = "";
+
+       ast_rwlock_rdlock(&mime_types_lock);
+       for (i = 0; i < mime_types_len; i++) {
+               if (ast_rtp_mime_types[i].payload_type.asterisk_format && asterisk_format && format &&
+                       (ast_format_cmp(format, ast_rtp_mime_types[i].payload_type.format) != AST_FORMAT_CMP_NOT_EQUAL)) {
+                       if ((ast_format_cmp(format, ast_format_g726_aal2) == AST_FORMAT_CMP_EQUAL) &&
+                                       (options & AST_RTP_OPT_G726_NONSTANDARD)) {
+                               res = "G726-32";
                                break;
-                       }
-                       continue;
-               }
-               /* Read in frame from channel */
-               fr = ast_read(who);
-               other = (who == c0) ? c1 : c0;
-               /* Depending on the frame we may need to break out of our bridge */
-               if (!fr || ((fr->frametype == AST_FRAME_DTMF_BEGIN || fr->frametype == AST_FRAME_DTMF_END) &&
-                           ((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) |
-                           ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)))) {
-                       /* Record received frame and who */
-                       *fo = fr;
-                       *rc = who;
-                       ast_debug(1, "rtp-engine-local-bridge: Ooh, got a %s\n", fr ? "digit" : "hangup");
-                       res = AST_BRIDGE_COMPLETE;
-                       break;
-               } else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
-                       if ((fr->subclass == AST_CONTROL_HOLD) ||
-                           (fr->subclass == AST_CONTROL_UNHOLD) ||
-                           (fr->subclass == AST_CONTROL_VIDUPDATE) ||
-                           (fr->subclass == AST_CONTROL_T38) ||
-                           (fr->subclass == AST_CONTROL_SRCUPDATE)) {
-                               /* If we are going on hold, then break callback mode and P2P bridging */
-                               if (fr->subclass == AST_CONTROL_HOLD) {
-                                       if (instance0->engine->local_bridge) {
-                                               instance0->engine->local_bridge(instance0, NULL);
-                                       }
-                                       if (instance1->engine->local_bridge) {
-                                               instance1->engine->local_bridge(instance1, NULL);
-                                       }
-                                       instance0->bridged = NULL;
-                                       instance1->bridged = NULL;
-                               } else if (fr->subclass == AST_CONTROL_UNHOLD) {
-                                       if (instance0->engine->local_bridge) {
-                                               instance0->engine->local_bridge(instance0, instance1);
-                                       }
-                                       if (instance1->engine->local_bridge) {
-                                               instance1->engine->local_bridge(instance1, instance0);
-                                       }
-                                       instance0->bridged = instance1;
-                                       instance1->bridged = instance0;
-                               }
-                               ast_indicate_data(other, fr->subclass, fr->data.ptr, fr->datalen);
-                               ast_frfree(fr);
                        } else {
-                               *fo = fr;
-                               *rc = who;
-                               ast_debug(1, "rtp-engine-local-bridge: Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass, who->name);
-                               res = AST_BRIDGE_COMPLETE;
+                               res = ast_rtp_mime_types[i].subtype;
                                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);
-                       }
+               } else if (!ast_rtp_mime_types[i].payload_type.asterisk_format && !asterisk_format &&
+                       ast_rtp_mime_types[i].payload_type.rtp_code == code) {
 
-                       ast_frfree(fr);
+                       res = ast_rtp_mime_types[i].subtype;
+                       break;
                }
-               /* Swap priority */
-               cs[2] = cs[0];
-               cs[0] = cs[1];
-               cs[1] = cs[2];
        }
+       ast_rwlock_unlock(&mime_types_lock);
 
-       /* 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);
-       }
+       return res;
+}
 
-       instance0->bridged = NULL;
-       instance1->bridged = NULL;
+unsigned int ast_rtp_lookup_sample_rate2(int asterisk_format,
+       const struct ast_format *format, int code)
+{
+       unsigned int i;
+       unsigned int res = 0;
 
-       ast_poll_channel_del(c0, c1);
+       ast_rwlock_rdlock(&mime_types_lock);
+       for (i = 0; i < mime_types_len; ++i) {
+               if (ast_rtp_mime_types[i].payload_type.asterisk_format && asterisk_format && format &&
+                       (ast_format_cmp(format, ast_rtp_mime_types[i].payload_type.format) != AST_FORMAT_CMP_NOT_EQUAL)) {
+                       res = ast_rtp_mime_types[i].sample_rate;
+                       break;
+               } else if (!ast_rtp_mime_types[i].payload_type.asterisk_format && !asterisk_format &&
+                       ast_rtp_mime_types[i].payload_type.rtp_code == code) {
+                       res = ast_rtp_mime_types[i].sample_rate;
+                       break;
+               }
+       }
+       ast_rwlock_unlock(&mime_types_lock);
 
        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)
+char *ast_rtp_lookup_mime_multiple2(struct ast_str *buf, struct ast_format_cap *ast_format_capability, int rtp_capability, const int asterisk_format, enum ast_rtp_options options)
 {
-       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);
+       int found = 0;
+       const char *name;
+       if (!buf) {
+               return NULL;
        }
 
-       /* 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);
+
+       if (asterisk_format) {
+               int x;
+               struct ast_format *tmp_fmt;
+               for (x = 0; x < ast_format_cap_count(ast_format_capability); x++) {
+                       tmp_fmt = ast_format_cap_get_format(ast_format_capability, x);
+                       name = ast_rtp_lookup_mime_subtype2(asterisk_format, tmp_fmt, 0, options);
+                       ao2_ref(tmp_fmt, -1);
+                       ast_str_append(&buf, 0, "%s|", name);
+                       found = 1;
                }
        } else {
-               ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c1->name, c0->name);
+               int x;
+               ast_str_append(&buf, 0, "0x%x (", (unsigned int) rtp_capability);
+               for (x = 1; x <= AST_RTP_MAX; x <<= 1) {
+                       if (rtp_capability & x) {
+                               name = ast_rtp_lookup_mime_subtype2(asterisk_format, NULL, x, options);
+                               ast_str_append(&buf, 0, "%s|", name);
+                               found = 1;
+                       }
+               }
        }
 
-       ast_channel_unlock(c0);
-       ast_channel_unlock(c1);
+       ast_str_append(&buf, 0, "%s", found ? ")" : "nothing)");
 
-       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;
-               }
+       return ast_str_buffer(buf);
+}
 
-               /* 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);
-               }
+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;
+}
 
-               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);
-               }
+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;
+}
 
-               if ((inaddrcmp(&t1, &ac1)) ||
-                   (vinstance1 && inaddrcmp(&vt1, &vac1)) ||
-                   (tinstance1 && inaddrcmp(&tt1, &tac1)) ||
-                   (codec1 != oldcodec1)) {
-                       ast_debug(1, "Oooh, '%s' changed end address to %s:%d (format %d)\n",
-                                 c1->name, ast_inet_ntoa(t1.sin_addr), ntohs(t1.sin_port), codec1);
-                       ast_debug(1, "Oooh, '%s' changed end vaddress to %s:%d (format %d)\n",
-                                 c1->name, ast_inet_ntoa(vt1.sin_addr), ntohs(vt1.sin_port), codec1);
-                       ast_debug(1, "Oooh, '%s' changed end taddress to %s:%d (format %d)\n",
-                                 c1->name, ast_inet_ntoa(tt1.sin_addr), ntohs(tt1.sin_port), codec1);
-                       ast_debug(1, "Oooh, '%s' was %s:%d/(format %d)\n",
-                                 c1->name, ast_inet_ntoa(ac1.sin_addr), ntohs(ac1.sin_port), oldcodec1);
-                       ast_debug(1, "Oooh, '%s' was %s:%d/(format %d)\n",
-                                 c1->name, ast_inet_ntoa(vac1.sin_addr), ntohs(vac1.sin_port), oldcodec1);
-                       ast_debug(1, "Oooh, '%s' was %s:%d/(format %d)\n",
-                                 c1->name, ast_inet_ntoa(tac1.sin_addr), ntohs(tac1.sin_port), oldcodec1);
-                       if (glue0->update_peer(c0, t1.sin_addr.s_addr ? instance1 : NULL, vt1.sin_addr.s_addr ? vinstance1 : NULL, tt1.sin_addr.s_addr ? tinstance1 : NULL, codec1, 0)) {
-                               ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c0->name, c1->name);
-                       }
-                       memcpy(&ac1, &t1, sizeof(ac1));
-                       memcpy(&vac1, &vt1, sizeof(vac1));
-                       memcpy(&tac1, &tt1, sizeof(tac1));
-                       oldcodec1 = codec1;
-               }
-               if ((inaddrcmp(&t0, &ac0)) ||
-                   (vinstance0 && inaddrcmp(&vt0, &vac0)) ||
-                   (tinstance0 && inaddrcmp(&tt0, &tac0))) {
-                       ast_debug(1, "Oooh, '%s' changed end address to %s:%d (format %d)\n",
-                                 c0->name, ast_inet_ntoa(t0.sin_addr), ntohs(t0.sin_port), codec0);
-                       ast_debug(1, "Oooh, '%s' was %s:%d/(format %d)\n",
-                                 c0->name, ast_inet_ntoa(ac0.sin_addr), ntohs(ac0.sin_port), oldcodec0);
-                       if (glue1->update_peer(c1, t0.sin_addr.s_addr ? instance0 : NULL, vt0.sin_addr.s_addr ? vinstance0 : NULL, tt0.sin_addr.s_addr ? tinstance0 : NULL, codec0, 0)) {
-                               ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c1->name, c0->name);
-                       }
-                       memcpy(&ac0, &t0, sizeof(ac0));
-                       memcpy(&vac0, &vt0, sizeof(vac0));
-                       memcpy(&tac0, &tt0, sizeof(tac0));
-                       oldcodec0 = codec0;
-               }
+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;
+}
 
-               /* Wait for frame to come in on the channels */
-               if (!(who = ast_waitfor_n(cs, 2, &timeoutms))) {
-                       if (!timeoutms) {
-                               res = AST_BRIDGE_RETRY;
-                               break;
-                       }
-                       ast_debug(1, "Ooh, empty read...\n");
-                       if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
-                               break;
-                       }
-                       continue;
-               }
-               fr = ast_read(who);
-               other = (who == c0) ? c1 : c0;
-               if (!fr || ((fr->frametype == AST_FRAME_DTMF_BEGIN || fr->frametype == AST_FRAME_DTMF_END) &&
-                           (((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) ||
-                            ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1))))) {
-                       /* Break out of bridge */
-                       *fo = fr;
-                       *rc = who;
-                       ast_debug(1, "Oooh, got a %s\n", fr ? "digit" : "hangup");
-                       res = AST_BRIDGE_COMPLETE;
-                       break;
-               } else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
-                       if ((fr->subclass == AST_CONTROL_HOLD) ||
-                           (fr->subclass == AST_CONTROL_UNHOLD) ||
-                           (fr->subclass == AST_CONTROL_VIDUPDATE) ||
-                           (fr->subclass == AST_CONTROL_T38) ||
-                           (fr->subclass == AST_CONTROL_SRCUPDATE)) {
-                               if (fr->subclass == AST_CONTROL_HOLD) {
-                                       /* If we someone went on hold we want the other side to reinvite back to us */
-                                       if (who == c0) {
-                                               glue1->update_peer(c1, NULL, NULL, NULL, 0, 0);
-                                       } else {
-                                               glue0->update_peer(c0, NULL, NULL, NULL, 0, 0);
-                                       }
-                               } else if (fr->subclass == AST_CONTROL_UNHOLD) {
-                                       /* If they went off hold they should go back to being direct */
-                                       if (who == c0) {
-                                               glue1->update_peer(c1, instance0, vinstance0, tinstance0, codec0, 0);
-                                       } else {
-                                               glue0->update_peer(c0, instance1, vinstance1, tinstance1, codec1, 0);
-                                       }
-                               }
-                               /* Update local address information */
-                               ast_rtp_instance_get_remote_address(instance0, &t0);
-                               memcpy(&ac0, &t0, sizeof(ac0));
-                               ast_rtp_instance_get_remote_address(instance1, &t1);
-                               memcpy(&ac1, &t1, sizeof(ac1));
-                               /* Update codec information */
-                               if (glue0->get_codec && c0->tech_pvt) {
-                                       oldcodec0 = codec0 = glue0->get_codec(c0);
-                               }
-                               if (glue1->get_codec && c1->tech_pvt) {
-                                       oldcodec1 = codec1 = glue1->get_codec(c1);
-                               }
-                               ast_indicate_data(other, fr->subclass, fr->data.ptr, fr->datalen);
-                               ast_frfree(fr);
-                       } else {
-                               *fo = fr;
-                               *rc = who;
-                               ast_debug(1, "Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass, who->name);
-                               return AST_BRIDGE_COMPLETE;
-                       }
-               } else {
-                       if ((fr->frametype == AST_FRAME_DTMF_BEGIN) ||
-                           (fr->frametype == AST_FRAME_DTMF_END) ||
-                           (fr->frametype == AST_FRAME_VOICE) ||
-                           (fr->frametype == AST_FRAME_VIDEO) ||
-                           (fr->frametype == AST_FRAME_IMAGE) ||
-                           (fr->frametype == AST_FRAME_HTML) ||
-                           (fr->frametype == AST_FRAME_MODEM) ||
-                           (fr->frametype == AST_FRAME_TEXT)) {
-                               ast_write(other, fr);
-                       }
-                       ast_frfree(fr);
-               }
-               /* Swap priority */
-               cs[2] = cs[0];
-               cs[0] = cs[1];
-               cs[1] = cs[2];
+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);
        }
+}
 
-       if (glue0->update_peer(c0, NULL, NULL, NULL, 0, 0)) {
-               ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
+void ast_rtp_instance_change_source(struct ast_rtp_instance *instance)
+{
+       if (instance->engine->change_source) {
+               instance->engine->change_source(instance);
        }
-       if (glue1->update_peer(c1, NULL, NULL, NULL, 0, 0)) {
-               ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
+}
+
+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;
+}
 
-       instance0->bridged = NULL;
-       instance1->bridged = NULL;
+struct ast_rtp_glue *ast_rtp_instance_get_glue(const char *type)
+{
+       struct ast_rtp_glue *glue = NULL;
 
-       ast_poll_channel_del(c0, c1);
+       AST_RWLIST_RDLOCK(&glues);
 
-       return res;
+       AST_RWLIST_TRAVERSE(&glues, glue, entry) {
+               if (!strcasecmp(glue->type, type)) {
+                       break;
+               }
+       }
+
+       AST_RWLIST_UNLOCK(&glues);
+
+       return glue;
 }
 
 /*!
@@ -1132,230 +1665,134 @@ static void unref_instance_cond(struct ast_rtp_instance **instance)
        }
 }
 
-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 *ast_rtp_instance_get_bridged(struct ast_rtp_instance *instance)
 {
-       struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
-                       *vinstance0 = NULL, *vinstance1 = NULL,
-                       *tinstance0 = NULL, *tinstance1 = NULL;
-       struct ast_rtp_glue *glue0, *glue1;
-       enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID, text_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
-       enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID, text_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
-       enum ast_bridge_result res = AST_BRIDGE_FAILED;
-       int codec0 = 0, codec1 = 0;
-       int unlock_chans = 1;
+       return instance->bridged;
+}
+
+void ast_rtp_instance_set_bridged(struct ast_rtp_instance *instance, struct ast_rtp_instance *bridged)
+{
+       instance->bridged = bridged;
+}
+
+void ast_rtp_instance_early_bridge_make_compatible(struct ast_channel *c_dst, struct ast_channel *c_src)
+{
+       struct ast_rtp_instance *instance_dst = NULL, *instance_src = NULL,
+               *vinstance_dst = NULL, *vinstance_src = NULL,
+               *tinstance_dst = NULL, *tinstance_src = NULL;
+       struct ast_rtp_glue *glue_dst, *glue_src;
+       enum ast_rtp_glue_result audio_glue_dst_res = AST_RTP_GLUE_RESULT_FORBID, video_glue_dst_res = AST_RTP_GLUE_RESULT_FORBID;
+       enum ast_rtp_glue_result audio_glue_src_res = AST_RTP_GLUE_RESULT_FORBID, video_glue_src_res = AST_RTP_GLUE_RESULT_FORBID;
+       struct ast_format_cap *cap_dst = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
+       struct ast_format_cap *cap_src = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
 
        /* 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);
-       }
+       ast_channel_lock_both(c_dst, c_src);
 
-       /* 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);
+       if (!cap_src || !cap_dst) {
                goto done;
        }
 
        /* Grab glue that binds each channel to something using the RTP engine */
-       if (!(glue0 = ast_rtp_instance_get_glue(c0->tech->type)) || !(glue1 = ast_rtp_instance_get_glue(c1->tech->type))) {
-               ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", glue0 ? c1->name : c0->name);
+       if (!(glue_dst = ast_rtp_instance_get_glue(ast_channel_tech(c_dst)->type)) || !(glue_src = ast_rtp_instance_get_glue(ast_channel_tech(c_src)->type))) {
+               ast_debug(1, "Can't find native functions for channel '%s'\n", glue_dst ? ast_channel_name(c_src) : ast_channel_name(c_dst));
                goto done;
        }
 
-       audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
-       video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
-       text_glue0_res = glue0->get_trtp_info ? glue0->get_trtp_info(c0, &tinstance0) : AST_RTP_GLUE_RESULT_FORBID;
+       audio_glue_dst_res = glue_dst->get_rtp_info(c_dst, &instance_dst);
+       video_glue_dst_res = glue_dst->get_vrtp_info ? glue_dst->get_vrtp_info(c_dst, &vinstance_dst) : AST_RTP_GLUE_RESULT_FORBID;
 
-       audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
-       video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
-       text_glue1_res = glue1->get_trtp_info ? glue1->get_trtp_info(c1, &tinstance1) : AST_RTP_GLUE_RESULT_FORBID;
+       audio_glue_src_res = glue_src->get_rtp_info(c_src, &instance_src);
+       video_glue_src_res = glue_src->get_vrtp_info ? glue_src->get_vrtp_info(c_src, &vinstance_src) : 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_glue_dst_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue_dst_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue_dst_res != AST_RTP_GLUE_RESULT_REMOTE)) {
+               audio_glue_dst_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 (video_glue_src_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue_src_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue_src_res != AST_RTP_GLUE_RESULT_REMOTE)) {
+               audio_glue_src_res = AST_RTP_GLUE_RESULT_FORBID;
+       }
+       if (audio_glue_dst_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue_dst_res == AST_RTP_GLUE_RESULT_FORBID || video_glue_dst_res == AST_RTP_GLUE_RESULT_REMOTE) && glue_dst->get_codec) {
+               glue_dst->get_codec(c_dst, cap_dst);
+       }
+       if (audio_glue_src_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue_src_res == AST_RTP_GLUE_RESULT_FORBID || video_glue_src_res == AST_RTP_GLUE_RESULT_REMOTE) && glue_src->get_codec) {
+               glue_src->get_codec(c_src, cap_src);
        }
 
        /* 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;
+       if (audio_glue_dst_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue_src_res != AST_RTP_GLUE_RESULT_REMOTE) {
                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;
+       /* Make sure we have matching codecs */
+       if (!ast_format_cap_iscompatible(cap_dst, cap_src)) {
                goto done;
        }
 
-       /* Make sure that codecs match */
-       codec0 = glue0->get_codec ? glue0->get_codec(c0) : 0;
-       codec1 = glue1->get_codec ? glue1->get_codec(c1) : 0;
-       if (codec0 && codec1 && !(codec0 & codec1)) {
-               ast_debug(1, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1);
-               res = AST_BRIDGE_FAILED_NOWARN;
-               goto done;
+       ast_rtp_codecs_payloads_xover(&instance_src->codecs, &instance_dst->codecs, instance_dst);
+
+       if (vinstance_dst && vinstance_src) {
+               ast_rtp_codecs_payloads_xover(&vinstance_src->codecs, &vinstance_dst->codecs, vinstance_dst);
+       }
+       if (tinstance_dst && tinstance_src) {
+               ast_rtp_codecs_payloads_xover(&tinstance_src->codecs, &tinstance_dst->codecs, tinstance_dst);
        }
 
-       /* Depending on the end result for bridging either do a local bridge or remote bridge */
-       if (audio_glue0_res == AST_RTP_GLUE_RESULT_LOCAL || audio_glue1_res == AST_RTP_GLUE_RESULT_LOCAL) {
-               ast_verbose(VERBOSE_PREFIX_3 "Locally bridging %s and %s\n", c0->name, c1->name);
-               res = local_bridge_loop(c0, c1, instance0, instance1, timeoutms, flags, fo, rc, c0->tech_pvt, c1->tech_pvt);
+       if (glue_dst->update_peer(c_dst, instance_src, vinstance_src, tinstance_src, cap_src, 0)) {
+               ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n",
+                       ast_channel_name(c_dst), ast_channel_name(c_src));
        } else {
-               ast_verbose(VERBOSE_PREFIX_3 "Remotely bridging %s and %s\n", c0->name, c1->name);
-               res = remote_bridge_loop(c0, c1, instance0, instance1, vinstance0, vinstance1,
-                               tinstance0, tinstance1, glue0, glue1, codec0, codec1, timeoutms, flags,
-                               fo, rc, c0->tech_pvt, c1->tech_pvt);
+               ast_debug(1, "Seeded SDP of '%s' with that of '%s'\n",
+                       ast_channel_name(c_dst), ast_channel_name(c_src));
        }
 
-       unlock_chans = 0;
-
 done:
-       if (unlock_chans) {
-               ast_channel_unlock(c0);
-               ast_channel_unlock(c1);
-       }
+       ast_channel_unlock(c_dst);
+       ast_channel_unlock(c_src);
 
-       unref_instance_cond(&instance0);
-       unref_instance_cond(&instance1);
-       unref_instance_cond(&vinstance0);
-       unref_instance_cond(&vinstance1);
-       unref_instance_cond(&tinstance0);
-       unref_instance_cond(&tinstance1);
+       ao2_cleanup(cap_dst);
+       ao2_cleanup(cap_src);
 
-       return res;
+       unref_instance_cond(&instance_dst);
+       unref_instance_cond(&instance_src);
+       unref_instance_cond(&vinstance_dst);
+       unref_instance_cond(&vinstance_src);
+       unref_instance_cond(&tinstance_dst);
+       unref_instance_cond(&tinstance_src);
 }
 
-struct ast_rtp_instance *ast_rtp_instance_get_bridged(struct ast_rtp_instance *instance)
-{
-       return instance->bridged;
-}
-
-void ast_rtp_instance_early_bridge_make_compatible(struct ast_channel *c0, struct ast_channel *c1)
-{
-       struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
-               *vinstance0 = NULL, *vinstance1 = NULL,
-               *tinstance0 = NULL, *tinstance1 = NULL;
-       struct ast_rtp_glue *glue0, *glue1;
-       enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID, text_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
-       enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID, text_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
-       int codec0 = 0, codec1 = 0;
-       int res = 0;
-
-       /* Lock both channels so we can look for the glue that binds them together */
-       ast_channel_lock(c0);
-       while (ast_channel_trylock(c1)) {
-               ast_channel_unlock(c0);
-               usleep(1);
-               ast_channel_lock(c0);
-       }
-
-       /* Grab glue that binds each channel to something using the RTP engine */
-       if (!(glue0 = ast_rtp_instance_get_glue(c0->tech->type)) || !(glue1 = ast_rtp_instance_get_glue(c1->tech->type))) {
-               ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", glue0 ? c1->name : c0->name);
-               goto done;
-       }
-
-       audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
-       video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
-       text_glue0_res = glue0->get_trtp_info ? glue0->get_trtp_info(c0, &tinstance0) : AST_RTP_GLUE_RESULT_FORBID;
-
-       audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
-       video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
-       text_glue1_res = glue1->get_trtp_info ? glue1->get_trtp_info(c1, &tinstance1) : AST_RTP_GLUE_RESULT_FORBID;
-
-       /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
-       if (video_glue0_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue0_res != AST_RTP_GLUE_RESULT_REMOTE)) {
-               audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
-       }
-       if (video_glue1_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue1_res != AST_RTP_GLUE_RESULT_REMOTE)) {
-               audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
-       }
-       if (audio_glue0_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue0_res == AST_RTP_GLUE_RESULT_FORBID || video_glue0_res == AST_RTP_GLUE_RESULT_REMOTE) && glue0->get_codec(c0)) {
-               codec0 = glue0->get_codec(c0);
-       }
-       if (audio_glue1_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue1_res == AST_RTP_GLUE_RESULT_FORBID || video_glue1_res == AST_RTP_GLUE_RESULT_REMOTE) && glue1->get_codec(c1)) {
-               codec1 = glue1->get_codec(c1);
-       }
-
-       /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
-       if (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE) {
-               goto done;
-       }
-
-       /* Make sure we have matching codecs */
-       if (!(codec0 & codec1)) {
-               goto done;
-       }
-
-       ast_rtp_codecs_payloads_copy(&instance0->codecs, &instance1->codecs, instance1);
-
-       if (vinstance0 && vinstance1) {
-               ast_rtp_codecs_payloads_copy(&vinstance0->codecs, &vinstance1->codecs, vinstance1);
-       }
-       if (tinstance0 && tinstance1) {
-               ast_rtp_codecs_payloads_copy(&tinstance0->codecs, &tinstance1->codecs, tinstance1);
-       }
-
-       res = 0;
-
-done:
-       ast_channel_unlock(c0);
-       ast_channel_unlock(c1);
-
-       unref_instance_cond(&instance0);
-       unref_instance_cond(&instance1);
-       unref_instance_cond(&vinstance0);
-       unref_instance_cond(&vinstance1);
-       unref_instance_cond(&tinstance0);
-       unref_instance_cond(&tinstance1);
-
-       if (!res) {
-               ast_debug(1, "Seeded SDP of '%s' with that of '%s'\n", c0->name, c1 ? c1->name : "<unspecified>");
-       }
-}
-
-int ast_rtp_instance_early_bridge(struct ast_channel *c0, struct ast_channel *c1)
+int ast_rtp_instance_early_bridge(struct ast_channel *c0, struct ast_channel *c1)
 {
        struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
                        *vinstance0 = NULL, *vinstance1 = NULL,
                        *tinstance0 = NULL, *tinstance1 = NULL;
        struct ast_rtp_glue *glue0, *glue1;
-       enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID, text_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
-       enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID, text_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
-       int codec0 = 0, codec1 = 0;
-       int res = 0;
+       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;
+       struct ast_format_cap *cap0 = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
+       struct ast_format_cap *cap1 = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
 
        /* If there is no second channel just immediately bail out, we are of no use in that scenario */
-       if (!c1) {
+       if (!c1 || !cap1 || !cap0) {
+               ao2_cleanup(cap0);
+               ao2_cleanup(cap1);
                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);
-       }
+       ast_channel_lock_both(c0, c1);
 
        /* 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);
+       if (!(glue0 = ast_rtp_instance_get_glue(ast_channel_tech(c0)->type)) || !(glue1 = ast_rtp_instance_get_glue(ast_channel_tech(c1)->type))) {
+               ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", glue0 ? ast_channel_name(c1) : ast_channel_name(c0));
                goto done;
        }
 
        audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
        video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
-       text_glue0_res = glue0->get_trtp_info ? glue0->get_trtp_info(c0, &tinstance0) : AST_RTP_GLUE_RESULT_FORBID;
 
        audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
        video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
-       text_glue1_res = glue1->get_trtp_info ? glue1->get_trtp_info(c1, &tinstance1) : AST_RTP_GLUE_RESULT_FORBID;
 
        /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
        if (video_glue0_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue0_res != AST_RTP_GLUE_RESULT_REMOTE)) {
@@ -1364,11 +1801,11 @@ int ast_rtp_instance_early_bridge(struct ast_channel *c0, struct ast_channel *c1
        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_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) {
+               glue0->get_codec(c0, cap0);
        }
-       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 (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) {
+               glue1->get_codec(c1, cap1);
        }
 
        /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
@@ -1377,21 +1814,22 @@ int ast_rtp_instance_early_bridge(struct ast_channel *c0, struct ast_channel *c1
        }
 
        /* Make sure we have matching codecs */
-       if (!(codec0 & codec1)) {
+       if (!ast_format_cap_iscompatible(cap0, cap1)) {
                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>");
+       if (glue0->update_peer(c0, instance1, vinstance1, tinstance1, cap1, 0)) {
+               ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", ast_channel_name(c0), c1 ? ast_channel_name(c1) : "<unspecified>");
        }
 
-       res = 0;
-
 done:
        ast_channel_unlock(c0);
        ast_channel_unlock(c1);
 
+       ao2_cleanup(cap0);
+       ao2_cleanup(cap1);
+
        unref_instance_cond(&instance0);
        unref_instance_cond(&instance1);
        unref_instance_cond(&vinstance0);
@@ -1399,11 +1837,9 @@ done:
        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>");
-       }
+       ast_debug(1, "Setting early bridge SDP of '%s' with that of '%s'\n", ast_channel_name(c0), c1 ? ast_channel_name(c1) : "<unspecified>");
 
-       return res;
+       return 0;
 }
 
 int ast_rtp_red_init(struct ast_rtp_instance *instance, int buffer_time, int *payloads, int generations)
@@ -1423,7 +1859,7 @@ int ast_rtp_instance_get_stats(struct ast_rtp_instance *instance, struct ast_rtp
 
 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;
+       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 */
@@ -1446,8 +1882,8 @@ char *ast_rtp_instance_get_quality(struct ast_rtp_instance *instance, enum ast_r
 
        /* Now actually fill the buffer with the good information */
        if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY) {
-               snprintf(buf, size, "ssrc=%i;themssrc=%u;lp=%u;rxjitter=%u;rxcount=%u;txjitter=%u;txcount=%u;rlp=%u;rtt=%u",
-                        stats.local_ssrc, stats.remote_ssrc, stats.rxploss, stats.txjitter, stats.rxcount, stats.rxjitter, stats.txcount, stats.txploss, stats.rtt);
+               snprintf(buf, size, "ssrc=%u;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.rxjitter, stats.rxcount, stats.txjitter, 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));
@@ -1463,44 +1899,70 @@ char *ast_rtp_instance_get_quality(struct ast_rtp_instance *instance, enum ast_r
 
 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);
+       char quality_buf[AST_MAX_USER_FIELD];
+       char *quality;
+       struct ast_channel *bridge;
+
+       bridge = ast_channel_bridge_peer(chan);
+       if (bridge) {
+               ast_channel_lock_both(chan, bridge);
+               ast_channel_stage_snapshot(bridge);
+       } else {
+               ast_channel_lock(chan);
+       }
+       ast_channel_stage_snapshot(chan);
 
-       if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
+       quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY,
+               quality_buf, sizeof(quality_buf));
+       if (quality) {
                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)))) {
+       quality = ast_rtp_instance_get_quality(instance,
+               AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, quality_buf, sizeof(quality_buf));
+       if (quality) {
                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)))) {
+       quality = ast_rtp_instance_get_quality(instance,
+               AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, quality_buf, sizeof(quality_buf));
+       if (quality) {
                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)))) {
+       quality = ast_rtp_instance_get_quality(instance,
+               AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, quality_buf, sizeof(quality_buf));
+       if (quality) {
                pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSRTT", quality);
                if (bridge) {
                        pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSRTTBRIDGED", quality);
                }
        }
+
+       ast_channel_stage_snapshot_done(chan);
+       ast_channel_unlock(chan);
+       if (bridge) {
+               ast_channel_stage_snapshot_done(bridge);
+               ast_channel_unlock(bridge);
+               ast_channel_unref(bridge);
+       }
 }
 
-int ast_rtp_instance_set_read_format(struct ast_rtp_instance *instance, int format)
+int ast_rtp_instance_set_read_format(struct ast_rtp_instance *instance, struct ast_format *format)
 {
        return instance->engine->set_read_format ? instance->engine->set_read_format(instance, format) : -1;
 }
 
-int ast_rtp_instance_set_write_format(struct ast_rtp_instance *instance, int format)
+int ast_rtp_instance_set_write_format(struct ast_rtp_instance *instance, struct ast_format *format)
 {
        return instance->engine->set_write_format ? instance->engine->set_write_format(instance, format) : -1;
 }
@@ -1517,16 +1979,21 @@ int ast_rtp_instance_make_compatible(struct ast_channel *chan, struct ast_rtp_in
 
        ast_channel_lock(peer);
 
-       if (!(glue = ast_rtp_instance_get_glue(peer->tech->type))) {
+       if (!(glue = ast_rtp_instance_get_glue(ast_channel_tech(peer)->type))) {
                ast_channel_unlock(peer);
                return -1;
        }
 
        glue->get_rtp_info(peer, &peer_instance);
-
-       if (!peer_instance || peer_instance->engine != instance->engine) {
+       if (!peer_instance) {
+               ast_log(LOG_ERROR, "Unable to get_rtp_info for peer type %s\n", glue->type);
+               ast_channel_unlock(peer);
+               return -1;
+       }
+       if (peer_instance->engine != instance->engine) {
+               ast_log(LOG_ERROR, "Peer engine mismatch for type %s\n", glue->type);
                ast_channel_unlock(peer);
-               peer_instance = (ao2_ref(peer_instance, -1), NULL);
+               ao2_ref(peer_instance, -1);
                return -1;
        }
 
@@ -1534,17 +2001,32 @@ int ast_rtp_instance_make_compatible(struct ast_channel *chan, struct ast_rtp_in
 
        ast_channel_unlock(peer);
 
-       peer_instance = (ao2_ref(peer_instance, -1), NULL);
+       ao2_ref(peer_instance, -1);
+       peer_instance = NULL;
 
        return res;
 }
 
+void ast_rtp_instance_available_formats(struct ast_rtp_instance *instance, struct ast_format_cap *to_endpoint, struct ast_format_cap *to_asterisk, struct ast_format_cap *result)
+{
+       if (instance->engine->available_formats) {
+               instance->engine->available_formats(instance, to_endpoint, to_asterisk, result);
+               if (ast_format_cap_count(result)) {
+                       return;
+               }
+       }
+
+       ast_translate_available_formats(to_endpoint, to_asterisk, result);
+}
+
 int ast_rtp_instance_activate(struct ast_rtp_instance *instance)
 {
        return instance->engine->activate ? instance->engine->activate(instance) : 0;
 }
 
-void ast_rtp_instance_stun_request(struct ast_rtp_instance *instance, struct sockaddr_in *suggestion, const char *username)
+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);
@@ -1561,6 +2043,11 @@ void ast_rtp_instance_set_hold_timeout(struct ast_rtp_instance *instance, int ti
        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;
@@ -1570,3 +2057,795 @@ 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;
+}
+
+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 *remote_policy, struct ast_srtp_policy *local_policy, int rtcp)
+{
+       int res = 0;
+       struct ast_srtp **srtp;
+
+       if (!res_srtp) {
+               return -1;
+       }
+
+
+       srtp = rtcp ? &instance->rtcp_srtp : &instance->srtp;
+
+       if (!*srtp) {
+               res = res_srtp->create(srtp, instance, remote_policy);
+       } else {
+               res = res_srtp->replace(srtp, instance, remote_policy);
+       }
+       if (!res) {
+               res = res_srtp->add_stream(*srtp, local_policy);
+       }
+
+       return res;
+}
+
+struct ast_srtp *ast_rtp_instance_get_srtp(struct ast_rtp_instance *instance, int rtcp)
+{
+       if (rtcp && instance->rtcp_srtp) {
+               return instance->rtcp_srtp;
+       }
+       else {
+               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;
+}
+
+struct ast_rtp_engine_ice *ast_rtp_instance_get_ice(struct ast_rtp_instance *instance)
+{
+       return instance->engine->ice;
+}
+
+struct ast_rtp_engine_dtls *ast_rtp_instance_get_dtls(struct ast_rtp_instance *instance)
+{
+       return instance->engine->dtls;
+}
+
+int ast_rtp_dtls_cfg_parse(struct ast_rtp_dtls_cfg *dtls_cfg, const char *name, const char *value)
+{
+       if (!strcasecmp(name, "dtlsenable")) {
+               dtls_cfg->enabled = ast_true(value) ? 1 : 0;
+       } else if (!strcasecmp(name, "dtlsverify")) {
+               if (!strcasecmp(value, "yes")) {
+                       dtls_cfg->verify = AST_RTP_DTLS_VERIFY_FINGERPRINT | AST_RTP_DTLS_VERIFY_CERTIFICATE;
+               } else if (!strcasecmp(value, "fingerprint")) {
+                       dtls_cfg->verify = AST_RTP_DTLS_VERIFY_FINGERPRINT;
+               } else if (!strcasecmp(value, "certificate")) {
+                       dtls_cfg->verify = AST_RTP_DTLS_VERIFY_CERTIFICATE;
+               } else if (!strcasecmp(value, "no")) {
+                       dtls_cfg->verify = AST_RTP_DTLS_VERIFY_NONE;
+               } else {
+                       return -1;
+               }
+       } else if (!strcasecmp(name, "dtlsrekey")) {
+               if (sscanf(value, "%30u", &dtls_cfg->rekey) != 1) {
+                       return -1;
+               }
+       } else if (!strcasecmp(name, "dtlscertfile")) {
+               ast_free(dtls_cfg->certfile);
+               if (!ast_strlen_zero(value) && !ast_file_is_readable(value)) {
+                       ast_log(LOG_ERROR, "%s file %s does not exist or is not readable\n", name, value);
+                       return -1;
+               }
+               dtls_cfg->certfile = ast_strdup(value);
+       } else if (!strcasecmp(name, "dtlsprivatekey")) {
+               ast_free(dtls_cfg->pvtfile);
+               if (!ast_strlen_zero(value) && !ast_file_is_readable(value)) {
+                       ast_log(LOG_ERROR, "%s file %s does not exist or is not readable\n", name, value);
+                       return -1;
+               }
+               dtls_cfg->pvtfile = ast_strdup(value);
+       } else if (!strcasecmp(name, "dtlscipher")) {
+               ast_free(dtls_cfg->cipher);
+               dtls_cfg->cipher = ast_strdup(value);
+       } else if (!strcasecmp(name, "dtlscafile")) {
+               ast_free(dtls_cfg->cafile);
+               if (!ast_strlen_zero(value) && !ast_file_is_readable(value)) {
+                       ast_log(LOG_ERROR, "%s file %s does not exist or is not readable\n", name, value);
+                       return -1;
+               }
+               dtls_cfg->cafile = ast_strdup(value);
+       } else if (!strcasecmp(name, "dtlscapath") || !strcasecmp(name, "dtlscadir")) {
+               ast_free(dtls_cfg->capath);
+               if (!ast_strlen_zero(value) && !ast_file_is_readable(value)) {
+                       ast_log(LOG_ERROR, "%s file %s does not exist or is not readable\n", name, value);
+                       return -1;
+               }
+               dtls_cfg->capath = ast_strdup(value);
+       } else if (!strcasecmp(name, "dtlssetup")) {
+               if (!strcasecmp(value, "active")) {
+                       dtls_cfg->default_setup = AST_RTP_DTLS_SETUP_ACTIVE;
+               } else if (!strcasecmp(value, "passive")) {
+                       dtls_cfg->default_setup = AST_RTP_DTLS_SETUP_PASSIVE;
+               } else if (!strcasecmp(value, "actpass")) {
+                       dtls_cfg->default_setup = AST_RTP_DTLS_SETUP_ACTPASS;
+               }
+       } else if (!strcasecmp(name, "dtlsfingerprint")) {
+               if (!strcasecmp(value, "sha-256")) {
+                       dtls_cfg->hash = AST_RTP_DTLS_HASH_SHA256;
+               } else if (!strcasecmp(value, "sha-1")) {
+                       dtls_cfg->hash = AST_RTP_DTLS_HASH_SHA1;
+               }
+       } else {
+               return -1;
+       }
+
+       return 0;
+}
+
+void ast_rtp_dtls_cfg_copy(const struct ast_rtp_dtls_cfg *src_cfg, struct ast_rtp_dtls_cfg *dst_cfg)
+{
+       ast_rtp_dtls_cfg_free(dst_cfg);         /* Prevent a double-call leaking memory via ast_strdup */
+
+       dst_cfg->enabled = src_cfg->enabled;
+       dst_cfg->verify = src_cfg->verify;
+       dst_cfg->rekey = src_cfg->rekey;
+       dst_cfg->suite = src_cfg->suite;
+       dst_cfg->hash = src_cfg->hash;
+       dst_cfg->certfile = ast_strdup(src_cfg->certfile);
+       dst_cfg->pvtfile = ast_strdup(src_cfg->pvtfile);
+       dst_cfg->cipher = ast_strdup(src_cfg->cipher);
+       dst_cfg->cafile = ast_strdup(src_cfg->cafile);
+       dst_cfg->capath = ast_strdup(src_cfg->capath);
+       dst_cfg->default_setup = src_cfg->default_setup;
+}
+
+void ast_rtp_dtls_cfg_free(struct ast_rtp_dtls_cfg *dtls_cfg)
+{
+       ast_free(dtls_cfg->certfile);
+       dtls_cfg->certfile = NULL;
+       ast_free(dtls_cfg->pvtfile);
+       dtls_cfg->pvtfile = NULL;
+       ast_free(dtls_cfg->cipher);
+       dtls_cfg->cipher = NULL;
+       ast_free(dtls_cfg->cafile);
+       dtls_cfg->cafile = NULL;
+       ast_free(dtls_cfg->capath);
+       dtls_cfg->capath = NULL;
+}
+
+/*! \internal
+ * \brief Small helper routine that cleans up entry i in
+ * \c ast_rtp_mime_types.
+ */
+static void rtp_engine_mime_type_cleanup(int i)
+{
+       ao2_cleanup(ast_rtp_mime_types[i].payload_type.format);
+       memset(&ast_rtp_mime_types[i], 0, sizeof(struct ast_rtp_mime_type));
+}
+
+static void set_next_mime_type(struct ast_format *format, int rtp_code, const char *type, const char *subtype, unsigned int sample_rate)
+{
+       int x;
+
+       ast_rwlock_wrlock(&mime_types_lock);
+
+       x = mime_types_len;
+       if (ARRAY_LEN(ast_rtp_mime_types) <= x) {
+               ast_rwlock_unlock(&mime_types_lock);
+               return;
+       }
+
+       /* Make sure any previous value in ast_rtp_mime_types is cleaned up */
+       memset(&ast_rtp_mime_types[x], 0, sizeof(struct ast_rtp_mime_type));    
+       if (format) {
+               ast_rtp_mime_types[x].payload_type.asterisk_format = 1;
+               ast_rtp_mime_types[x].payload_type.format = ao2_bump(format);
+       } else {
+               ast_rtp_mime_types[x].payload_type.rtp_code = rtp_code;
+       }
+       ast_copy_string(ast_rtp_mime_types[x].type, type, sizeof(ast_rtp_mime_types[x].type));
+       ast_copy_string(ast_rtp_mime_types[x].subtype, subtype, sizeof(ast_rtp_mime_types[x].subtype));
+       ast_rtp_mime_types[x].sample_rate = sample_rate;
+       mime_types_len++;
+
+       ast_rwlock_unlock(&mime_types_lock);
+}
+
+static void add_static_payload(int map, struct ast_format *format, int rtp_code)
+{
+       int x;
+       struct ast_rtp_payload_type *type;
+
+       /*
+        * ARRAY_LEN's result is cast to an int so 'map' is not autocast to a size_t,
+        * which if negative would cause an assertion.
+        */
+       ast_assert(map < (int)ARRAY_LEN(static_RTP_PT));
+
+       ast_rwlock_wrlock(&static_RTP_PT_lock);
+       if (map < 0) {
+               /* find next available dynamic payload slot */
+               for (x = AST_RTP_PT_FIRST_DYNAMIC; x < AST_RTP_MAX_PT; ++x) {
+                       if (!static_RTP_PT[x]) {
+                               map = x;
+                               break;
+                       }
+               }
+
+               /* http://www.iana.org/assignments/rtp-parameters
+                * RFC 3551, Section 3: "[...] applications which need to define more
+                * than 32 dynamic payload types MAY bind codes below 96, in which case
+                * it is RECOMMENDED that unassigned payload type numbers be used
+                * first". Updated by RFC 5761, Section 4: "[...] values in the range
+                * 64-95 MUST NOT be used [to avoid conflicts with RTCP]". Summaries:
+                * https://tools.ietf.org/html/draft-roach-mmusic-unified-plan#section-3.2.1.2
+                * https://tools.ietf.org/html/draft-wu-avtcore-dynamic-pt-usage#section-3
+                */
+               if (map < 0) {
+                       for (x = MAX(ast_option_rtpptdynamic, 35); x <= AST_RTP_PT_LAST_REASSIGN; ++x) {
+                               if (!static_RTP_PT[x]) {
+                                       map = x;
+                                       break;
+                               }
+                       }
+               }
+               /* Yet, reusing mappings below 35 is not supported in Asterisk because
+                * when Compact Headers are activated, no rtpmap is send for those below
+                * 35. If you want to use 35 and below
+                * A) do not use Compact Headers,
+                * B) remove that code in chan_sip/res_pjsip, or
+                * C) add a flag that this RTP Payload Type got reassigned dynamically
+                *    and requires a rtpmap even with Compact Headers enabled.
+                */
+               if (map < 0) {
+                       for (x = MAX(ast_option_rtpptdynamic, 20); x < 35; ++x) {
+                               if (!static_RTP_PT[x]) {
+                                       map = x;
+                                       break;
+                               }
+                       }
+               }
+               if (map < 0) {
+                       for (x = MAX(ast_option_rtpptdynamic, 0); x < 20; ++x) {
+                               if (!static_RTP_PT[x]) {
+                                       map = x;
+                                       break;
+                               }
+                       }
+               }
+
+               if (map < 0) {
+                       if (format) {
+                               ast_log(LOG_WARNING, "No Dynamic RTP mapping available for format %s\n",
+                                       ast_format_get_name(format));
+                       } else {
+                               ast_log(LOG_WARNING, "No Dynamic RTP mapping available for RTP code %d\n",
+                                       rtp_code);
+                       }
+                       ast_rwlock_unlock(&static_RTP_PT_lock);
+                       return;
+               }
+       }
+
+       type = ast_rtp_engine_alloc_payload_type();
+       if (type) {
+               if (format) {
+                       ao2_ref(format, +1);
+                       type->format = format;
+                       type->asterisk_format = 1;
+               } else {
+                       type->rtp_code = rtp_code;
+               }
+               type->payload = map;
+               type->primary_mapping = 1;
+               ao2_cleanup(static_RTP_PT[map]);
+               static_RTP_PT[map] = type;
+       }
+       ast_rwlock_unlock(&static_RTP_PT_lock);
+}
+
+int ast_rtp_engine_load_format(struct ast_format *format)
+{
+       set_next_mime_type(format,
+               0,
+               ast_codec_media_type2str(ast_format_get_type(format)),
+               ast_format_get_codec_name(format),
+               ast_format_get_sample_rate(format));
+       add_static_payload(-1, format, 0);
+
+       return 0;
+}
+
+int ast_rtp_engine_unload_format(struct ast_format *format)
+{
+       int x;
+       int y = 0;
+
+       ast_rwlock_wrlock(&static_RTP_PT_lock);
+       /* remove everything pertaining to this format id from the lists */
+       for (x = 0; x < AST_RTP_MAX_PT; x++) {
+               if (static_RTP_PT[x]
+                       && ast_format_cmp(static_RTP_PT[x]->format, format) == AST_FORMAT_CMP_EQUAL) {
+                       ao2_ref(static_RTP_PT[x], -1);
+                       static_RTP_PT[x] = NULL;
+               }
+       }
+       ast_rwlock_unlock(&static_RTP_PT_lock);
+
+       ast_rwlock_wrlock(&mime_types_lock);
+       /* rebuild the list skipping the items matching this id */
+       for (x = 0; x < mime_types_len; x++) {
+               if (ast_format_cmp(ast_rtp_mime_types[x].payload_type.format, format) == AST_FORMAT_CMP_EQUAL) {
+                       rtp_engine_mime_type_cleanup(x);
+                       continue;
+               }
+               if (x != y) {
+                       ast_rtp_mime_types[y] = ast_rtp_mime_types[x];
+               }
+               y++;
+       }
+       mime_types_len = y;
+       ast_rwlock_unlock(&mime_types_lock);
+       return 0;
+}
+
+/*!
+ * \internal
+ * \brief \ref stasis message payload for RTCP messages
+ */
+struct rtcp_message_payload {
+       struct ast_channel_snapshot *snapshot;  /*< The channel snapshot, if available */
+       struct ast_rtp_rtcp_report *report;     /*< The RTCP report */
+       struct ast_json *blob;                  /*< Extra JSON data to publish */
+};
+
+static void rtcp_message_payload_dtor(void *obj)
+{
+       struct rtcp_message_payload *payload = obj;
+
+       ao2_cleanup(payload->report);
+       ao2_cleanup(payload->snapshot);
+       ast_json_unref(payload->blob);
+}
+
+static struct ast_manager_event_blob *rtcp_report_to_ami(struct stasis_message *msg)
+{
+       struct rtcp_message_payload *payload = stasis_message_data(msg);
+       RAII_VAR(struct ast_str *, channel_string, NULL, ast_free);
+       RAII_VAR(struct ast_str *, packet_string, ast_str_create(512), ast_free);
+       unsigned int ssrc = payload->report->ssrc;
+       unsigned int type = payload->report->type;
+       unsigned int report_count = payload->report->reception_report_count;
+       int i;
+
+       if (!packet_string) {
+               return NULL;
+       }
+
+       if (payload->snapshot) {
+               channel_string = ast_manager_build_channel_state_string(payload->snapshot);
+               if (!channel_string) {
+                       return NULL;
+               }
+       }
+
+       if (payload->blob) {
+               /* Optional data */
+               struct ast_json *to = ast_json_object_get(payload->blob, "to");
+               struct ast_json *from = ast_json_object_get(payload->blob, "from");
+               struct ast_json *rtt = ast_json_object_get(payload->blob, "rtt");
+               if (to) {
+                       ast_str_append(&packet_string, 0, "To: %s\r\n", ast_json_string_get(to));
+               }
+               if (from) {
+                       ast_str_append(&packet_string, 0, "From: %s\r\n", ast_json_string_get(from));
+               }
+               if (rtt) {
+                       ast_str_append(&packet_string, 0, "RTT: %4.4f\r\n", ast_json_real_get(rtt));
+               }
+       }
+
+       ast_str_append(&packet_string, 0, "SSRC: 0x%.8x\r\n", ssrc);
+       ast_str_append(&packet_string, 0, "PT: %u(%s)\r\n", type, type== AST_RTP_RTCP_SR ? "SR" : "RR");
+       ast_str_append(&packet_string, 0, "ReportCount: %u\r\n", report_count);
+       if (type == AST_RTP_RTCP_SR) {
+               ast_str_append(&packet_string, 0, "SentNTP: %lu.%06lu\r\n",
+                       (unsigned long)payload->report->sender_information.ntp_timestamp.tv_sec,
+                       (unsigned long)payload->report->sender_information.ntp_timestamp.tv_usec);
+               ast_str_append(&packet_string, 0, "SentRTP: %u\r\n",
+                               payload->report->sender_information.rtp_timestamp);
+               ast_str_append(&packet_string, 0, "SentPackets: %u\r\n",
+                               payload->report->sender_information.packet_count);
+               ast_str_append(&packet_string, 0, "SentOctets: %u\r\n",
+                               payload->report->sender_information.octet_count);
+       }
+
+       for (i = 0; i < report_count; i++) {
+               RAII_VAR(struct ast_str *, report_string, NULL, ast_free);
+
+               if (!payload->report->report_block[i]) {
+                       break;
+               }
+
+               report_string = ast_str_create(256);
+               if (!report_string) {
+                       return NULL;
+               }
+
+               ast_str_append(&report_string, 0, "Report%dSourceSSRC: 0x%.8x\r\n",
+                               i, payload->report->report_block[i]->source_ssrc);
+               ast_str_append(&report_string, 0, "Report%dFractionLost: %d\r\n",
+                               i, payload->report->report_block[i]->lost_count.fraction);
+               ast_str_append(&report_string, 0, "Report%dCumulativeLost: %u\r\n",
+                               i, payload->report->report_block[i]->lost_count.packets);
+               ast_str_append(&report_string, 0, "Report%dHighestSequence: %u\r\n",
+                               i, payload->report->report_block[i]->highest_seq_no & 0xffff);
+               ast_str_append(&report_string, 0, "Report%dSequenceNumberCycles: %u\r\n",
+                               i, payload->report->report_block[i]->highest_seq_no >> 16);
+               ast_str_append(&report_string, 0, "Report%dIAJitter: %u\r\n",
+                               i, payload->report->report_block[i]->ia_jitter);
+               ast_str_append(&report_string, 0, "Report%dLSR: %u\r\n",
+                               i, payload->report->report_block[i]->lsr);
+               ast_str_append(&report_string, 0, "Report%dDLSR: %4.4f\r\n",
+                               i, ((double)payload->report->report_block[i]->dlsr) / 65536);
+               ast_str_append(&packet_string, 0, "%s", ast_str_buffer(report_string));
+       }
+
+       return ast_manager_event_blob_create(EVENT_FLAG_REPORTING,
+               stasis_message_type(msg) == ast_rtp_rtcp_received_type() ? "RTCPReceived" : "RTCPSent",
+               "%s%s",
+               AS_OR(channel_string, ""),
+               ast_str_buffer(packet_string));
+}
+
+static struct ast_json *rtcp_report_to_json(struct stasis_message *msg,
+       const struct stasis_message_sanitizer *sanitize)
+{
+       struct rtcp_message_payload *payload = stasis_message_data(msg);
+       RAII_VAR(struct ast_json *, json_rtcp_report, NULL, ast_json_unref);
+       RAII_VAR(struct ast_json *, json_rtcp_report_blocks, NULL, ast_json_unref);
+       RAII_VAR(struct ast_json *, json_rtcp_sender_info, NULL, ast_json_unref);
+       RAII_VAR(struct ast_json *, json_channel, NULL, ast_json_unref);
+       int i;
+
+       json_rtcp_report_blocks = ast_json_array_create();
+       if (!json_rtcp_report_blocks) {
+               return NULL;
+       }
+
+       for (i = 0; i < payload->report->reception_report_count && payload->report->report_block[i]; i++) {
+               struct ast_json *json_report_block;
+               char str_lsr[32];
+               snprintf(str_lsr, sizeof(str_lsr), "%u", payload->report->report_block[i]->lsr);
+               json_report_block = ast_json_pack("{s: i, s: i, s: i, s: i, s: i, s: s, s: i}",
+                               "source_ssrc", payload->report->report_block[i]->source_ssrc,
+                               "fraction_lost", payload->report->report_block[i]->lost_count.fraction,
+                               "packets_lost", payload->report->report_block[i]->lost_count.packets,
+                               "highest_seq_no", payload->report->report_block[i]->highest_seq_no,
+                               "ia_jitter", payload->report->report_block[i]->ia_jitter,
+                               "lsr", str_lsr,
+                               "dlsr", payload->report->report_block[i]->dlsr);
+               if (!json_report_block) {
+                       return NULL;
+               }
+
+               if (ast_json_array_append(json_rtcp_report_blocks, json_report_block)) {
+                       return NULL;
+               }
+       }
+
+       if (payload->report->type == AST_RTP_RTCP_SR) {
+               char sec[32];
+               char usec[32];
+               snprintf(sec, sizeof(sec), "%lu", (unsigned long)payload->report->sender_information.ntp_timestamp.tv_sec);
+               snprintf(usec, sizeof(usec), "%lu", (unsigned long)payload->report->sender_information.ntp_timestamp.tv_usec);
+               json_rtcp_sender_info = ast_json_pack("{s: s, s: s, s: i, s: i, s: i}",
+                               "ntp_timestamp_sec", sec,
+                               "ntp_timestamp_usec", usec,
+                               "rtp_timestamp", payload->report->sender_information.rtp_timestamp,
+                               "packets", payload->report->sender_information.packet_count,
+                               "octets", payload->report->sender_information.octet_count);
+               if (!json_rtcp_sender_info) {
+                       return NULL;
+               }
+       }
+
+       json_rtcp_report = ast_json_pack("{s: i, s: i, s: i, s: o, s: o}",
+                       "ssrc", payload->report->ssrc,
+                       "type", payload->report->type,
+                       "report_count", payload->report->reception_report_count,
+                       "sender_information", json_rtcp_sender_info ? ast_json_ref(json_rtcp_sender_info) : ast_json_ref(ast_json_null()),
+                       "report_blocks", ast_json_ref(json_rtcp_report_blocks));
+       if (!json_rtcp_report) {
+               return NULL;
+       }
+
+       if (payload->snapshot) {
+               json_channel = ast_channel_snapshot_to_json(payload->snapshot, sanitize);
+               if (!json_channel) {
+                       return NULL;
+               }
+       }
+
+       return ast_json_pack("{s: o, s: o, s: o}",
+               "channel", payload->snapshot ? ast_json_ref(json_channel) : ast_json_ref(ast_json_null()),
+               "rtcp_report", ast_json_ref(json_rtcp_report),
+               "blob", ast_json_deep_copy(payload->blob));
+}
+
+static void rtp_rtcp_report_dtor(void *obj)
+{
+       int i;
+       struct ast_rtp_rtcp_report *rtcp_report = obj;
+
+       for (i = 0; i < rtcp_report->reception_report_count; i++) {
+               ast_free(rtcp_report->report_block[i]);
+       }
+}
+
+struct ast_rtp_rtcp_report *ast_rtp_rtcp_report_alloc(unsigned int report_blocks)
+{
+       struct ast_rtp_rtcp_report *rtcp_report;
+
+       /* Size of object is sizeof the report + the number of report_blocks * sizeof pointer */
+       rtcp_report = ao2_alloc((sizeof(*rtcp_report) + report_blocks * sizeof(struct ast_rtp_rtcp_report_block *)),
+               rtp_rtcp_report_dtor);
+
+       return rtcp_report;
+}
+
+void ast_rtp_publish_rtcp_message(struct ast_rtp_instance *rtp,
+               struct stasis_message_type *message_type,
+               struct ast_rtp_rtcp_report *report,
+               struct ast_json *blob)
+{
+       RAII_VAR(struct rtcp_message_payload *, payload, NULL, ao2_cleanup);
+       RAII_VAR(struct stasis_message *, message, NULL, ao2_cleanup);
+
+       if (!message_type) {
+               return;
+       }
+
+       payload = ao2_alloc(sizeof(*payload), rtcp_message_payload_dtor);
+       if (!payload || !report) {
+               return;
+       }
+
+       if (!ast_strlen_zero(rtp->channel_uniqueid)) {
+               payload->snapshot = ast_channel_snapshot_get_latest(rtp->channel_uniqueid);
+       }
+       if (blob) {
+               payload->blob = blob;
+               ast_json_ref(blob);
+       }
+       ao2_ref(report, +1);
+       payload->report = report;
+
+       message = stasis_message_create(message_type, payload);
+       if (!message) {
+               return;
+       }
+
+       stasis_publish(ast_rtp_topic(), message);
+}
+
+/*!
+ * @{ \brief Define RTCP/RTP message types.
+ */
+STASIS_MESSAGE_TYPE_DEFN(ast_rtp_rtcp_sent_type,
+               .to_ami = rtcp_report_to_ami,
+               .to_json = rtcp_report_to_json,);
+STASIS_MESSAGE_TYPE_DEFN(ast_rtp_rtcp_received_type,
+               .to_ami = rtcp_report_to_ami,
+               .to_json = rtcp_report_to_json,);
+/*! @} */
+
+struct stasis_topic *ast_rtp_topic(void)
+{
+       return rtp_topic;
+}
+
+static void rtp_engine_shutdown(void)
+{
+       int x;
+
+       ao2_cleanup(rtp_topic);
+       rtp_topic = NULL;
+       STASIS_MESSAGE_TYPE_CLEANUP(ast_rtp_rtcp_received_type);
+       STASIS_MESSAGE_TYPE_CLEANUP(ast_rtp_rtcp_sent_type);
+
+       ast_rwlock_wrlock(&static_RTP_PT_lock);
+       for (x = 0; x < AST_RTP_MAX_PT; x++) {
+               ao2_cleanup(static_RTP_PT[x]);
+               static_RTP_PT[x] = NULL;
+       }
+       ast_rwlock_unlock(&static_RTP_PT_lock);
+
+       ast_rwlock_wrlock(&mime_types_lock);
+       for (x = 0; x < mime_types_len; x++) {
+               if (ast_rtp_mime_types[x].payload_type.format) {
+                       rtp_engine_mime_type_cleanup(x);
+               }
+       }
+       mime_types_len = 0;
+       ast_rwlock_unlock(&mime_types_lock);
+}
+
+int ast_rtp_engine_init(void)
+{
+       ast_rwlock_init(&mime_types_lock);
+       ast_rwlock_init(&static_RTP_PT_lock);
+
+       rtp_topic = stasis_topic_create("rtp_topic");
+       if (!rtp_topic) {
+               return -1;
+       }
+       STASIS_MESSAGE_TYPE_INIT(ast_rtp_rtcp_sent_type);
+       STASIS_MESSAGE_TYPE_INIT(ast_rtp_rtcp_received_type);
+       ast_register_cleanup(rtp_engine_shutdown);
+
+       /* Define all the RTP mime types available */
+       set_next_mime_type(ast_format_g723, 0, "audio", "G723", 8000);
+       set_next_mime_type(ast_format_gsm, 0, "audio", "GSM", 8000);
+       set_next_mime_type(ast_format_ulaw, 0, "audio", "PCMU", 8000);
+       set_next_mime_type(ast_format_ulaw, 0, "audio", "G711U", 8000);
+       set_next_mime_type(ast_format_alaw, 0, "audio", "PCMA", 8000);
+       set_next_mime_type(ast_format_alaw, 0, "audio", "G711A", 8000);
+       set_next_mime_type(ast_format_g726, 0, "audio", "G726-32", 8000);
+       set_next_mime_type(ast_format_adpcm, 0, "audio", "DVI4", 8000);
+       set_next_mime_type(ast_format_slin, 0, "audio", "L16", 8000);
+       set_next_mime_type(ast_format_slin16, 0, "audio", "L16", 16000);
+       set_next_mime_type(ast_format_slin16, 0, "audio", "L16-256", 16000);
+       set_next_mime_type(ast_format_slin12, 0, "audio", "L16", 12000);
+       set_next_mime_type(ast_format_slin24, 0, "audio", "L16", 24000);
+       set_next_mime_type(ast_format_slin32, 0, "audio", "L16", 32000);
+       set_next_mime_type(ast_format_slin44, 0, "audio", "L16", 44000);
+       set_next_mime_type(ast_format_slin48, 0, "audio", "L16", 48000);
+       set_next_mime_type(ast_format_slin96, 0, "audio", "L16", 96000);
+       set_next_mime_type(ast_format_slin192, 0, "audio", "L16", 192000);
+       set_next_mime_type(ast_format_lpc10, 0, "audio", "LPC", 8000);
+       set_next_mime_type(ast_format_g729, 0, "audio", "G729", 8000);
+       set_next_mime_type(ast_format_g729, 0, "audio", "G729A", 8000);
+       set_next_mime_type(ast_format_g729, 0, "audio", "G.729", 8000);
+       set_next_mime_type(ast_format_speex, 0, "audio", "speex", 8000);
+       set_next_mime_type(ast_format_speex16, 0,  "audio", "speex", 16000);
+       set_next_mime_type(ast_format_speex32, 0,  "audio", "speex", 32000);
+       set_next_mime_type(ast_format_ilbc, 0, "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 */
+       set_next_mime_type(ast_format_g722, 0, "audio", "G722", 8000);
+       set_next_mime_type(ast_format_g726_aal2, 0, "audio", "AAL2-G726-32", 8000);
+       set_next_mime_type(NULL, AST_RTP_DTMF, "audio", "telephone-event", 8000);
+       set_next_mime_type(NULL, AST_RTP_CISCO_DTMF, "audio", "cisco-telephone-event", 8000);
+       set_next_mime_type(NULL, AST_RTP_CN, "audio", "CN", 8000);
+       set_next_mime_type(ast_format_jpeg, 0, "video", "JPEG", 90000);
+       set_next_mime_type(ast_format_png, 0, "video", "PNG", 90000);
+       set_next_mime_type(ast_format_h261, 0, "video", "H261", 90000);
+       set_next_mime_type(ast_format_h263, 0, "video", "H263", 90000);
+       set_next_mime_type(ast_format_h263p, 0, "video", "h263-1998", 90000);
+       set_next_mime_type(ast_format_h264, 0, "video", "H264", 90000);
+       set_next_mime_type(ast_format_mp4, 0, "video", "MP4V-ES", 90000);
+       set_next_mime_type(ast_format_t140_red, 0, "text", "RED", 1000);
+       set_next_mime_type(ast_format_t140, 0, "text", "T140", 1000);
+       set_next_mime_type(ast_format_siren7, 0, "audio", "G7221", 16000);
+       set_next_mime_type(ast_format_siren14, 0, "audio", "G7221", 32000);
+       set_next_mime_type(ast_format_g719, 0, "audio", "G719", 48000);
+       /* Opus and VP8 */
+       set_next_mime_type(ast_format_opus, 0,  "audio", "opus", 48000);
+       set_next_mime_type(ast_format_vp8, 0,  "video", "VP8", 90000);
+
+       /* Define the static rtp payload mappings */
+       add_static_payload(0, ast_format_ulaw, 0);
+       #ifdef USE_DEPRECATED_G726
+       add_static_payload(2, ast_format_g726, 0);/* Technically this is G.721, but if Cisco can do it, so can we... */
+       #endif
+       add_static_payload(3, ast_format_gsm, 0);
+       add_static_payload(4, ast_format_g723, 0);
+       add_static_payload(5, ast_format_adpcm, 0);/* 8 kHz */
+       add_static_payload(6, ast_format_adpcm, 0); /* 16 kHz */
+       add_static_payload(7, ast_format_lpc10, 0);
+       add_static_payload(8, ast_format_alaw, 0);
+       add_static_payload(9, ast_format_g722, 0);
+       add_static_payload(10, ast_format_slin, 0); /* 2 channels */
+       add_static_payload(11, ast_format_slin, 0); /* 1 channel */
+       add_static_payload(13, NULL, AST_RTP_CN);
+       add_static_payload(16, ast_format_adpcm, 0); /* 11.025 kHz */
+       add_static_payload(17, ast_format_adpcm, 0); /* 22.050 kHz */
+       add_static_payload(18, ast_format_g729, 0);
+       add_static_payload(19, NULL, AST_RTP_CN);         /* Also used for CN */
+       add_static_payload(26, ast_format_jpeg, 0);
+       add_static_payload(31, ast_format_h261, 0);
+       add_static_payload(34, ast_format_h263, 0);
+       add_static_payload(97, ast_format_ilbc, 0);
+       add_static_payload(98, ast_format_h263p, 0);
+       add_static_payload(99, ast_format_h264, 0);
+       add_static_payload(101, NULL, AST_RTP_DTMF);
+       add_static_payload(102, ast_format_siren7, 0);
+       add_static_payload(103, ast_format_h263p, 0);
+       add_static_payload(104, ast_format_mp4, 0);
+       add_static_payload(105, ast_format_t140_red, 0);   /* Real time text chat (with redundancy encoding) */
+       add_static_payload(106, ast_format_t140, 0);     /* Real time text chat */
+       add_static_payload(110, ast_format_speex, 0);
+       add_static_payload(111, ast_format_g726, 0);
+       add_static_payload(112, ast_format_g726_aal2, 0);
+       add_static_payload(115, ast_format_siren14, 0);
+       add_static_payload(116, ast_format_g719, 0);
+       add_static_payload(117, ast_format_speex16, 0);
+       add_static_payload(118, ast_format_slin16, 0); /* 16 Khz signed linear */
+       add_static_payload(119, ast_format_speex32, 0);
+       add_static_payload(121, NULL, AST_RTP_CISCO_DTMF);   /* Must be type 121 */
+       add_static_payload(122, ast_format_slin12, 0);
+       add_static_payload(123, ast_format_slin24, 0);
+       add_static_payload(124, ast_format_slin32, 0);
+       add_static_payload(125, ast_format_slin44, 0);
+       add_static_payload(126, ast_format_slin48, 0);
+       add_static_payload(127, ast_format_slin96, 0);
+       /* payload types above 127 are not valid */
+       add_static_payload(96, ast_format_slin192, 0);
+       /* Opus and VP8 */
+       add_static_payload(100, ast_format_vp8, 0);
+       add_static_payload(107, ast_format_opus, 0);
+
+       return 0;
+}
+
+time_t ast_rtp_instance_get_last_tx(const struct ast_rtp_instance *rtp)
+{
+       return rtp->last_tx;
+}
+
+void ast_rtp_instance_set_last_tx(struct ast_rtp_instance *rtp, time_t time)
+{
+       rtp->last_tx = time;
+}
+
+time_t ast_rtp_instance_get_last_rx(const struct ast_rtp_instance *rtp)
+{
+       return rtp->last_rx;
+}
+
+void ast_rtp_instance_set_last_rx(struct ast_rtp_instance *rtp, time_t time)
+{
+       rtp->last_rx = time;
+}