ARI: Add ability to raise arbitrary User Events
[asterisk/asterisk.git] / main / rtp_engine.c
index 85ee9b6..5174b9c 100644 (file)
  * \author Joshua Colp <jcolp@digium.com>
  */
 
+/*** 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>
+               </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>
+               </managerEventInstance>
+       </managerEvent>
+ ***/
+
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
@@ -37,6 +149,16 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/options.h"
 #include "asterisk/astobj2.h"
 #include "asterisk/pbx.h"
+#include "asterisk/translate.h"
+#include "asterisk/netsock2.h"
+#include "asterisk/_private.h"
+#include "asterisk/framehook.h"
+#include "asterisk/stasis.h"
+#include "asterisk/json.h"
+#include "asterisk/stasis_channels.h"
+
+struct ast_srtp_res *res_srtp = NULL;
+struct ast_srtp_policy_res *res_srtp_policy = NULL;
 
 /*! Structure that represents an RTP session (instance) */
 struct ast_rtp_instance {
@@ -47,9 +169,9 @@ 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;
        /*! Address that we are sending RTP to */
-       struct sockaddr_in remote_address;
+       struct ast_sockaddr remote_address;
        /*! Instance that we are bridged to if doing remote or local bridging */
        struct ast_rtp_instance *bridged;
        /*! Payload and packetization information */
@@ -58,8 +180,14 @@ 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;
+       /*! Channel unique ID */
+       char channel_uniqueid[AST_MAX_UNIQUEID];
 };
 
 /*! List of RTP engines that are currently registered */
@@ -70,47 +198,14 @@ static AST_RWLIST_HEAD_STATIC(glues, ast_rtp_glue);
 
 /*! The following array defines the MIME Media type (and subtype) for each
    of our codecs, or RTP-specific data type. */
-static const struct ast_rtp_mime_type {
+static struct ast_rtp_mime_type {
        struct ast_rtp_payload_type payload_type;
        char *type;
        char *subtype;
        unsigned int sample_rate;
-} ast_rtp_mime_types[] = {
-       {{1, AST_FORMAT_G723_1}, "audio", "G723", 8000},
-       {{1, AST_FORMAT_GSM}, "audio", "GSM", 8000},
-       {{1, AST_FORMAT_ULAW}, "audio", "PCMU", 8000},
-       {{1, AST_FORMAT_ULAW}, "audio", "G711U", 8000},
-       {{1, AST_FORMAT_ALAW}, "audio", "PCMA", 8000},
-       {{1, AST_FORMAT_ALAW}, "audio", "G711A", 8000},
-       {{1, AST_FORMAT_G726}, "audio", "G726-32", 8000},
-       {{1, AST_FORMAT_ADPCM}, "audio", "DVI4", 8000},
-       {{1, AST_FORMAT_SLINEAR}, "audio", "L16", 8000},
-       {{1, AST_FORMAT_LPC10}, "audio", "LPC", 8000},
-       {{1, AST_FORMAT_G729A}, "audio", "G729", 8000},
-       {{1, AST_FORMAT_G729A}, "audio", "G729A", 8000},
-       {{1, AST_FORMAT_G729A}, "audio", "G.729", 8000},
-       {{1, AST_FORMAT_SPEEX}, "audio", "speex", 8000},
-       {{1, AST_FORMAT_ILBC}, "audio", "iLBC", 8000},
-       /* this is the sample rate listed in the RTP profile for the G.722
-                     codec, *NOT* the actual sample rate of the media stream
-       */
-       {{1, AST_FORMAT_G722}, "audio", "G722", 8000},
-       {{1, AST_FORMAT_G726_AAL2}, "audio", "AAL2-G726-32", 8000},
-       {{0, AST_RTP_DTMF}, "audio", "telephone-event", 8000},
-       {{0, AST_RTP_CISCO_DTMF}, "audio", "cisco-telephone-event", 8000},
-       {{0, AST_RTP_CN}, "audio", "CN", 8000},
-       {{1, AST_FORMAT_JPEG}, "video", "JPEG", 90000},
-       {{1, AST_FORMAT_PNG}, "video", "PNG", 90000},
-       {{1, AST_FORMAT_H261}, "video", "H261", 90000},
-       {{1, AST_FORMAT_H263}, "video", "H263", 90000},
-       {{1, AST_FORMAT_H263_PLUS}, "video", "h263-1998", 90000},
-       {{1, AST_FORMAT_H264}, "video", "H264", 90000},
-       {{1, AST_FORMAT_MP4_VIDEO}, "video", "MP4V-ES", 90000},
-       {{1, AST_FORMAT_T140RED}, "text", "RED", 1000},
-       {{1, AST_FORMAT_T140}, "text", "T140", 1000},
-       {{1, AST_FORMAT_SIREN7}, "audio", "G7221", 16000},
-       {{1, AST_FORMAT_SIREN14}, "audio", "G7221", 32000},
-};
+} 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 +217,11 @@ 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;
 
 int ast_rtp_engine_register2(struct ast_rtp_engine *engine, struct ast_module *module)
 {
@@ -263,6 +326,12 @@ static void instance_destructor(void *obj)
                return;
        }
 
+       if (instance->srtp) {
+               res_srtp->destroy(instance->srtp);
+       }
+
+       ast_rtp_codecs_payloads_destroy(&instance->codecs);
+
        /* Drop our engine reference */
        ast_module_unref(instance->engine->mod);
 
@@ -276,9 +345,11 @@ 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 sockaddr_in address = { 0, };
+       struct ast_sockaddr address = {{0,}};
        struct ast_rtp_instance *instance = NULL;
        struct ast_rtp_engine *engine = NULL;
 
@@ -313,10 +384,13 @@ struct ast_rtp_instance *ast_rtp_instance_new(const char *engine_name, struct sc
                return NULL;
        }
        instance->engine = engine;
-       instance->local_address.sin_family = AF_INET;
-       instance->local_address.sin_addr = sin->sin_addr;
-       instance->remote_address.sin_family = AF_INET;
-       address.sin_addr = sin->sin_addr;
+       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);
 
@@ -332,6 +406,16 @@ 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;
@@ -352,17 +436,17 @@ 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)
 {
-       instance->local_address.sin_addr = address->sin_addr;
-       instance->local_address.sin_port = address->sin_port;
+       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_remote_address(struct ast_rtp_instance *instance,
+               const struct ast_sockaddr *address)
 {
-       instance->remote_address.sin_addr = address->sin_addr;
-       instance->remote_address.sin_port = address->sin_port;
+       ast_sockaddr_copy(&instance->remote_address, address);
 
        /* moo */
 
@@ -373,30 +457,40 @@ int ast_rtp_instance_set_remote_address(struct ast_rtp_instance *instance, struc
        return 0;
 }
 
-int ast_rtp_instance_get_local_address(struct ast_rtp_instance *instance, struct sockaddr_in *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_remote_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->remote_address) != 0) {
+               ast_sockaddr_copy(address, &instance->remote_address);
                return 1;
        }
 
        return 0;
 }
 
+void ast_rtp_instance_get_remote_address(struct ast_rtp_instance *instance,
+               struct ast_sockaddr *address)
+{
+       ast_sockaddr_copy(address, &instance->remote_address);
+}
+
 void ast_rtp_instance_set_extended_prop(struct ast_rtp_instance *instance, int property, void *value)
 {
        if (instance->engine->extended_prop_set) {
@@ -432,66 +526,151 @@ struct ast_rtp_codecs *ast_rtp_instance_get_codecs(struct ast_rtp_instance *inst
        return &instance->codecs;
 }
 
+static int rtp_payload_type_hash(const void *obj, const int flags)
+{
+       const struct ast_rtp_payload_type *type = obj;
+       const int *payload = obj;
+
+       return (flags & OBJ_KEY) ? *payload : type->payload;
+}
+
+static int rtp_payload_type_cmp(void *obj, void *arg, int flags)
+{
+       struct ast_rtp_payload_type *type1 = obj, *type2 = arg;
+       const int *payload = arg;
+
+       return (type1->payload == (OBJ_KEY ? *payload : type2->payload)) ? CMP_MATCH | CMP_STOP : 0;
+}
+
+int ast_rtp_codecs_payloads_initialize(struct ast_rtp_codecs *codecs)
+{
+       if (!(codecs->payloads = ao2_container_alloc(AST_RTP_MAX_PT, rtp_payload_type_hash, rtp_payload_type_cmp))) {
+               return -1;
+       }
+
+       return 0;
+}
+
+void ast_rtp_codecs_payloads_destroy(struct ast_rtp_codecs *codecs)
+{
+       ao2_cleanup(codecs->payloads);
+}
+
 void ast_rtp_codecs_payloads_clear(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance)
 {
-       int i;
+       ast_rtp_codecs_payloads_destroy(codecs);
 
-       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);
+       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);
                }
        }
+
+       ast_rtp_codecs_payloads_initialize(codecs);
 }
 
 void ast_rtp_codecs_payloads_default(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance)
 {
        int i;
 
+       ast_rwlock_rdlock(&static_RTP_PT_lock);
        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 (static_RTP_PT[i].rtp_code || static_RTP_PT[i].asterisk_format) {
+                       struct ast_rtp_payload_type *type;
+
+                       if (!(type = ao2_alloc(sizeof(*type), NULL))) {
+                               /* Unfortunately if this occurs the payloads container will not contain all possible default payloads
+                                * but we err on the side of doing what we can in the hopes that the extreme memory conditions which
+                                * caused this to occur will go away.
+                                */
+                               continue;
+                       }
+
+                       type->payload = i;
+                       type->asterisk_format = static_RTP_PT[i].asterisk_format;
+                       type->rtp_code = static_RTP_PT[i].rtp_code;
+                       ast_format_copy(&type->format, &static_RTP_PT[i].format);
+
+                       ao2_link_flags(codecs->payloads, type, OBJ_NOLOCK);
+
                        if (instance && instance->engine && instance->engine->payload_set) {
-                               instance->engine->payload_set(instance, i, codecs->payloads[i].asterisk_format, codecs->payloads[i].code);
+                               instance->engine->payload_set(instance, i, type->asterisk_format, &type->format, type->rtp_code);
                        }
+
+                       ao2_ref(type, -1);
                }
        }
+       ast_rwlock_unlock(&static_RTP_PT_lock);
 }
 
 void ast_rtp_codecs_payloads_copy(struct ast_rtp_codecs *src, struct ast_rtp_codecs *dest, struct ast_rtp_instance *instance)
 {
        int i;
+       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);
-                       }
+               struct ast_rtp_payload_type *new_type;
+
+               if (!(type = ao2_find(src->payloads, &i, OBJ_KEY | OBJ_NOLOCK))) {
+                       continue;
+               }
+
+               if (!(new_type = ao2_alloc(sizeof(*new_type), NULL))) {
+                       continue;
+               }
+
+               ast_debug(2, "Copying payload %d from %p to %p\n", i, src, dest);
+
+               new_type->payload = i;
+               *new_type = *type;
+
+               ao2_link_flags(dest->payloads, new_type, OBJ_NOLOCK);
+
+               ao2_ref(new_type, -1);
+
+               if (instance && instance->engine && instance->engine->payload_set) {
+                       instance->engine->payload_set(instance, i, type->asterisk_format, &type->format, type->rtp_code);
                }
+
+               ao2_ref(type, -1);
        }
 }
 
 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 *type;
+
+       ast_rwlock_rdlock(&static_RTP_PT_lock);
+
+       if (payload < 0 || payload >= AST_RTP_MAX_PT) {
+               ast_rwlock_unlock(&static_RTP_PT_lock);
                return;
        }
 
-       codecs->payloads[payload].asterisk_format = static_RTP_PT[payload].asterisk_format;
-       codecs->payloads[payload].code = static_RTP_PT[payload].code;
+       if (!(type = ao2_find(codecs->payloads, &payload, OBJ_KEY | OBJ_NOLOCK))) {
+               if (!(type = ao2_alloc(sizeof(*type), NULL))) {
+                       ast_rwlock_unlock(&static_RTP_PT_lock);
+                       return;
+               }
+               type->payload = payload;
+               ao2_link_flags(codecs->payloads, type, OBJ_NOLOCK);
+       }
+
+       type->asterisk_format = static_RTP_PT[payload].asterisk_format;
+       type->rtp_code = static_RTP_PT[payload].rtp_code;
+       type->payload = payload;
+       ast_format_copy(&type->format, &static_RTP_PT[payload].format);
 
        ast_debug(1, "Setting payload %d based on m type on %p\n", payload, codecs);
 
        if (instance && instance->engine && instance->engine->payload_set) {
-               instance->engine->payload_set(instance, payload, codecs->payloads[payload].asterisk_format, codecs->payloads[payload].code);
+               instance->engine->payload_set(instance, payload, type->asterisk_format, &type->format, type->rtp_code);
        }
+
+       ao2_ref(type, -1);
+
+       ast_rwlock_unlock(&static_RTP_PT_lock);
 }
 
 int ast_rtp_codecs_payloads_set_rtpmap_type_rate(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int pt,
@@ -502,11 +681,13 @@ int ast_rtp_codecs_payloads_set_rtpmap_type_rate(struct ast_rtp_codecs *codecs,
        unsigned int i;
        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) {
+       ast_rwlock_rdlock(&mime_types_lock);
+       for (i = 0; i < mime_types_len; ++i) {
                const struct ast_rtp_mime_type *t = &ast_rtp_mime_types[i];
+               struct ast_rtp_payload_type *type;
 
                if (strcasecmp(mimesubtype, t->subtype)) {
                        continue;
@@ -517,28 +698,39 @@ 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;
+               if (!(type = ao2_find(codecs->payloads, &pt, OBJ_KEY | OBJ_NOLOCK))) {
+                       if (!(type = ao2_alloc(sizeof(*type), NULL))) {
+                               continue;
+                       }
+                       type->payload = pt;
+                       ao2_link_flags(codecs->payloads, type, OBJ_NOLOCK);
+               }
+
+               *type = t->payload_type;
+               type->payload = pt;
+
+               if ((t->payload_type.format.id == AST_FORMAT_G726) && t->payload_type.asterisk_format && (options & AST_RTP_OPT_G726_NONSTANDARD)) {
+                       ast_format_set(&type->format, AST_FORMAT_G726_AAL2, 0);
                }
 
                if (instance && instance->engine && instance->engine->payload_set) {
-                       instance->engine->payload_set(instance, pt, codecs->payloads[i].asterisk_format, codecs->payloads[i].code);
+                       instance->engine->payload_set(instance, pt, type->asterisk_format, &type->format, type->rtp_code);
                }
 
+               ao2_ref(type, -1);
+
                break;
        }
+       ast_rwlock_unlock(&mime_types_lock);
 
        return (found ? 0 : -2);
 }
@@ -550,122 +742,236 @@ 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) {
+       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;
+       ao2_find(codecs->payloads, &payload, OBJ_KEY | OBJ_NOLOCK | OBJ_NODATA | OBJ_UNLINK);
 
        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);
        }
 }
 
 struct ast_rtp_payload_type ast_rtp_codecs_payload_lookup(struct ast_rtp_codecs *codecs, int payload)
 {
-       struct ast_rtp_payload_type result = { .asterisk_format = 0, };
+       struct ast_rtp_payload_type result = { .asterisk_format = 0, }, *type;
 
-       if (payload < 0 || payload > AST_RTP_MAX_PT) {
+       if (payload < 0 || payload >= AST_RTP_MAX_PT) {
                return result;
        }
 
-       result.asterisk_format = codecs->payloads[payload].asterisk_format;
-       result.code = codecs->payloads[payload].code;
+       if ((type = ao2_find(codecs->payloads, &payload, OBJ_KEY | OBJ_NOLOCK))) {
+               result = *type;
+               ao2_ref(type, -1);
+       }
 
-       if (!result.code) {
+       if (!result.rtp_code && !result.asterisk_format) {
+               ast_rwlock_rdlock(&static_RTP_PT_lock);
                result = static_RTP_PT[payload];
+               ast_rwlock_unlock(&static_RTP_PT_lock);
        }
 
        return result;
 }
 
-void ast_rtp_codecs_payload_formats(struct ast_rtp_codecs *codecs, int *astformats, int *nonastformats)
+
+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;
 
-       *astformats = *nonastformats = 0;
+       if (payload < 0 || payload >= AST_RTP_MAX_PT) {
+               return NULL;
+       }
 
-       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;
-               }
+       if (!(type = ao2_find(codecs->payloads, &payload, OBJ_KEY | OBJ_NOLOCK))) {
+               return NULL;
        }
+
+       format = type->asterisk_format ? &type->format : NULL;
+
+       ao2_ref(type, -1);
+
+       return format;
 }
 
-int ast_rtp_codecs_payload_code(struct ast_rtp_codecs *codecs, const int asterisk_format, const int code)
+static int rtp_payload_type_add_ast(void *obj, void *arg, int flags)
 {
-       int i;
+       struct ast_rtp_payload_type *type = obj;
+       struct ast_format_cap *astformats = arg;
 
-       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 (type->asterisk_format) {
+               ast_format_cap_add(astformats, &type->format);
+       }
+
+       return 0;
+}
+
+static int rtp_payload_type_add_nonast(void *obj, void *arg, int flags)
+{
+       struct ast_rtp_payload_type *type = obj;
+       int *nonastformats = arg;
+
+       if (!type->asterisk_format) {
+               *nonastformats |= type->rtp_code;
+       }
+
+       return 0;
+}
+
+void ast_rtp_codecs_payload_formats(struct ast_rtp_codecs *codecs, struct ast_format_cap *astformats, int *nonastformats)
+{
+       ast_format_cap_remove_all(astformats);
+       *nonastformats = 0;
+
+       ao2_callback(codecs->payloads, OBJ_NODATA | OBJ_MULTIPLE | OBJ_NOLOCK, rtp_payload_type_add_ast, astformats);
+       ao2_callback(codecs->payloads, OBJ_NODATA | OBJ_MULTIPLE | OBJ_NOLOCK, rtp_payload_type_add_nonast, nonastformats);
+}
+
+static int rtp_payload_type_find_format(void *obj, void *arg, int flags)
+{
+       struct ast_rtp_payload_type *type = obj;
+       struct ast_format *format = arg;
+
+       return (type->asterisk_format && (ast_format_cmp(&type->format, format) != AST_FORMAT_CMP_NOT_EQUAL)) ? CMP_MATCH | CMP_STOP : 0;
+}
+
+static int rtp_payload_type_find_nonast_format(void *obj, void *arg, int flags)
+{
+       struct ast_rtp_payload_type *type = obj;
+       int *rtp_code = arg;
+
+       return ((!type->asterisk_format && (type->rtp_code == *rtp_code)) ? CMP_MATCH | CMP_STOP : 0);
+}
+
+int ast_rtp_codecs_payload_code(struct ast_rtp_codecs *codecs, int asterisk_format, const struct ast_format *format, int code)
+{
+       struct ast_rtp_payload_type *type;
+       int i, res = -1;
+
+       if (asterisk_format && format && (type = ao2_callback(codecs->payloads, OBJ_NOLOCK, rtp_payload_type_find_format, (void*)format))) {
+               res = type->payload;
+               ao2_ref(type, -1);
+               return res;
+       } else if (!asterisk_format && (type = ao2_callback(codecs->payloads, OBJ_NOLOCK, rtp_payload_type_find_nonast_format, (void*)&code))) {
+               res = type->payload;
+               ao2_ref(type, -1);
+               return res;
        }
 
+       ast_rwlock_rdlock(&static_RTP_PT_lock);
        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;
+               if (static_RTP_PT[i].asterisk_format && asterisk_format && format &&
+                       (ast_format_cmp(format, &static_RTP_PT[i].format) != AST_FORMAT_CMP_NOT_EQUAL)) {
+                       res = i;
+                       break;
+               } else if (!static_RTP_PT[i].asterisk_format && !asterisk_format &&
+                       (static_RTP_PT[i].rtp_code == code)) {
+                       res = i;
+                       break;
                }
        }
+       ast_rwlock_unlock(&static_RTP_PT_lock);
 
-       return -1;
+       return res;
 }
+int ast_rtp_codecs_find_payload_code(struct ast_rtp_codecs *codecs, int code)
+{
+       struct ast_rtp_payload_type *type;
+       int res = -1;
+
+       /* Search the payload type in the codecs passed */
+       if ((type = ao2_find(codecs->payloads, &code, OBJ_NOLOCK | OBJ_KEY)))
+       {
+               res = type->payload;
+               ao2_ref(type, -1);
+               return res;
+       }
 
-const char *ast_rtp_lookup_mime_subtype2(const int asterisk_format, const int code, enum ast_rtp_options options)
+       return res;
+}
+const char *ast_rtp_lookup_mime_subtype2(const int asterisk_format, struct ast_format *format, int code, enum ast_rtp_options options)
 {
        int i;
-
-       for (i = 0; i < ARRAY_LEN(ast_rtp_mime_types); i++) {
-               if (ast_rtp_mime_types[i].payload_type.code == code && ast_rtp_mime_types[i].payload_type.asterisk_format == asterisk_format) {
-                       if (asterisk_format && (code == AST_FORMAT_G726_AAL2) && (options & AST_RTP_OPT_G726_NONSTANDARD)) {
-                               return "G726-32";
+       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 ((format->id == AST_FORMAT_G726_AAL2) && (options & AST_RTP_OPT_G726_NONSTANDARD)) {
+                               res = "G726-32";
+                               break;
                        } else {
-                               return ast_rtp_mime_types[i].subtype;
+                               res = ast_rtp_mime_types[i].subtype;
+                               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].subtype;
+                       break;
                }
        }
+       ast_rwlock_unlock(&mime_types_lock);
 
-       return "";
+       return res;
 }
 
-unsigned int ast_rtp_lookup_sample_rate2(int asterisk_format, int code)
+unsigned int ast_rtp_lookup_sample_rate2(int asterisk_format, struct ast_format *format, int code)
 {
        unsigned int i;
+       unsigned int res = 0;
 
-       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(&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 0;
+       return res;
 }
 
-char *ast_rtp_lookup_mime_multiple2(struct ast_str *buf, const int capability, const int asterisk_format, enum ast_rtp_options options)
+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)
 {
-       int format, found = 0;
-
+       int found = 0;
+       const char *name;
        if (!buf) {
                return NULL;
        }
 
-       ast_str_append(&buf, 0, "0x%x (", capability);
 
-       for (format = 1; format < AST_RTP_MAX; format <<= 1) {
-               if (capability & format) {
-                       const char *name = ast_rtp_lookup_mime_subtype2(asterisk_format, format, options);
+       if (asterisk_format) {
+               struct ast_format tmp_fmt;
+               ast_format_cap_iter_start(ast_format_capability);
+               while (!ast_format_cap_iter_next(ast_format_capability, &tmp_fmt)) {
+                       name = ast_rtp_lookup_mime_subtype2(asterisk_format, &tmp_fmt, 0, options);
                        ast_str_append(&buf, 0, "%s|", name);
                        found = 1;
                }
+               ast_format_cap_iter_end(ast_format_capability);
+
+       } else {
+               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_str_append(&buf, 0, "%s", found ? ")" : "nothing)");
@@ -691,27 +997,32 @@ int ast_rtp_instance_dtmf_end(struct ast_rtp_instance *instance, char digit)
 {
        return instance->engine->dtmf_end ? instance->engine->dtmf_end(instance, digit) : -1;
 }
+int ast_rtp_instance_dtmf_end_with_duration(struct ast_rtp_instance *instance, char digit, unsigned int duration)
+{
+       return instance->engine->dtmf_end_with_duration ? instance->engine->dtmf_end_with_duration(instance, digit, duration) : -1;
+}
 
 int ast_rtp_instance_dtmf_mode_set(struct ast_rtp_instance *instance, enum ast_rtp_dtmf_mode dtmf_mode)
 {
-       if (!instance->engine->dtmf_mode_set || instance->engine->dtmf_mode_set(instance, dtmf_mode)) {
-               return -1;
-       }
-
-       instance->dtmf_mode = dtmf_mode;
-
-       return 0;
+       return (!instance->engine->dtmf_mode_set || instance->engine->dtmf_mode_set(instance, dtmf_mode)) ? -1 : 0;
 }
 
 enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get(struct ast_rtp_instance *instance)
 {
-       return instance->dtmf_mode;
+       return instance->engine->dtmf_mode_get ? instance->engine->dtmf_mode_get(instance) : 0;
+}
+
+void ast_rtp_instance_update_source(struct ast_rtp_instance *instance)
+{
+       if (instance->engine->update_source) {
+               instance->engine->update_source(instance);
+       }
 }
 
-void ast_rtp_instance_new_source(struct ast_rtp_instance *instance)
+void ast_rtp_instance_change_source(struct ast_rtp_instance *instance)
 {
-       if (instance->engine->new_source) {
-               instance->engine->new_source(instance);
+       if (instance->engine->change_source) {
+               instance->engine->change_source(instance);
        }
 }
 
@@ -749,589 +1060,111 @@ struct ast_rtp_glue *ast_rtp_instance_get_glue(const char *type)
        return glue;
 }
 
-static enum ast_bridge_result local_bridge_loop(struct ast_channel *c0, struct ast_channel *c1, struct ast_rtp_instance *instance0, struct ast_rtp_instance *instance1, int timeoutms, int flags, struct ast_frame **fo, struct ast_channel **rc, void *pvt0, void *pvt1)
+/*!
+ * \brief Conditionally unref an rtp instance
+ */
+static void unref_instance_cond(struct ast_rtp_instance **instance)
 {
-       enum ast_bridge_result res = AST_BRIDGE_FAILED;
-       struct ast_channel *who = NULL, *other = NULL, *cs[3] = { NULL, };
-       struct ast_frame *fr = NULL;
-
-       /* Start locally bridging both instances */
-       if (instance0->engine->local_bridge && instance0->engine->local_bridge(instance0, instance1)) {
-               ast_debug(1, "Failed to locally bridge %s to %s, backing out.\n", c0->name, c1->name);
-               ast_channel_unlock(c0);
-               ast_channel_unlock(c1);
-               return AST_BRIDGE_FAILED_NOWARN;
-       }
-       if (instance1->engine->local_bridge && instance1->engine->local_bridge(instance1, instance0)) {
-               ast_debug(1, "Failed to locally bridge %s to %s, backing out.\n", c1->name, c0->name);
-               if (instance0->engine->local_bridge) {
-                       instance0->engine->local_bridge(instance0, NULL);
-               }
-               ast_channel_unlock(c0);
-               ast_channel_unlock(c1);
-               return AST_BRIDGE_FAILED_NOWARN;
+       if (*instance) {
+               ao2_ref(*instance, -1);
+               *instance = NULL;
        }
+}
 
-       ast_channel_unlock(c0);
-       ast_channel_unlock(c1);
+struct ast_rtp_instance *ast_rtp_instance_get_bridged(struct ast_rtp_instance *instance)
+{
+       return instance->bridged;
+}
 
-       instance0->bridged = instance1;
-       instance1->bridged = instance0;
+void ast_rtp_instance_set_bridged(struct ast_rtp_instance *instance, struct ast_rtp_instance *bridged)
+{
+       instance->bridged = bridged;
+}
 
-       ast_poll_channel_add(c0, c1);
+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_NOLOCK);
+       struct ast_format_cap *cap_src = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK);
 
-       /* Hop into a loop waiting for a frame from either channel */
-       cs[0] = c0;
-       cs[1] = c1;
-       cs[2] = NULL;
-       for (;;) {
-               /* If the underlying formats have changed force this bridge to break */
-               if ((c0->rawreadformat != c1->rawwriteformat) || (c1->rawreadformat != c0->rawwriteformat)) {
-                       ast_debug(1, "rtp-engine-local-bridge: Oooh, formats changed, backing out\n");
-                       res = AST_BRIDGE_FAILED_NOWARN;
-                       break;
-               }
-               /* Check if anything changed */
-               if ((c0->tech_pvt != pvt0) ||
-                   (c1->tech_pvt != pvt1) ||
-                   (c0->masq || c0->masqr || c1->masq || c1->masqr) ||
-                   (c0->monitor || c0->audiohooks || c1->monitor || c1->audiohooks)) {
-                       ast_debug(1, "rtp-engine-local-bridge: Oooh, something is weird, backing out\n");
-                       /* If a masquerade needs to happen we have to try to read in a frame so that it actually happens. Without this we risk being called again and going into a loop */
-                       if ((c0->masq || c0->masqr) && (fr = ast_read(c0))) {
-                               ast_frfree(fr);
-                       }
-                       if ((c1->masq || c1->masqr) && (fr = ast_read(c1))) {
-                               ast_frfree(fr);
-                       }
-                       res = AST_BRIDGE_RETRY;
-                       break;
-               }
-               /* Wait on a channel to feed us a frame */
-               if (!(who = ast_waitfor_n(cs, 2, &timeoutms))) {
-                       if (!timeoutms) {
-                               res = AST_BRIDGE_RETRY;
-                               break;
-                       }
-                       ast_debug(2, "rtp-engine-local-bridge: Ooh, empty read...\n");
-                       if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
-                               break;
-                       }
-                       continue;
-               }
-               /* Read in frame from channel */
-               fr = ast_read(who);
-               other = (who == c0) ? c1 : c0;
-               /* Depending on the frame we may need to break out of our bridge */
-               if (!fr || ((fr->frametype == AST_FRAME_DTMF_BEGIN || fr->frametype == AST_FRAME_DTMF_END) &&
-                           ((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) |
-                           ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)))) {
-                       /* Record received frame and who */
-                       *fo = fr;
-                       *rc = who;
-                       ast_debug(1, "rtp-engine-local-bridge: Ooh, got a %s\n", fr ? "digit" : "hangup");
-                       res = AST_BRIDGE_COMPLETE;
-                       break;
-               } else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
-                       if ((fr->subclass == AST_CONTROL_HOLD) ||
-                           (fr->subclass == AST_CONTROL_UNHOLD) ||
-                           (fr->subclass == AST_CONTROL_VIDUPDATE) ||
-                           (fr->subclass == AST_CONTROL_T38) ||
-                           (fr->subclass == AST_CONTROL_SRCUPDATE)) {
-                               /* If we are going on hold, then break callback mode and P2P bridging */
-                               if (fr->subclass == AST_CONTROL_HOLD) {
-                                       if (instance0->engine->local_bridge) {
-                                               instance0->engine->local_bridge(instance0, NULL);
-                                       }
-                                       if (instance1->engine->local_bridge) {
-                                               instance1->engine->local_bridge(instance1, NULL);
-                                       }
-                                       instance0->bridged = NULL;
-                                       instance1->bridged = NULL;
-                               } else if (fr->subclass == AST_CONTROL_UNHOLD) {
-                                       if (instance0->engine->local_bridge) {
-                                               instance0->engine->local_bridge(instance0, instance1);
-                                       }
-                                       if (instance1->engine->local_bridge) {
-                                               instance1->engine->local_bridge(instance1, instance0);
-                                       }
-                                       instance0->bridged = instance1;
-                                       instance1->bridged = instance0;
-                               }
-                               ast_indicate_data(other, fr->subclass, fr->data.ptr, fr->datalen);
-                               ast_frfree(fr);
-                       } else {
-                               *fo = fr;
-                               *rc = who;
-                               ast_debug(1, "rtp-engine-local-bridge: Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass, who->name);
-                               res = AST_BRIDGE_COMPLETE;
-                               break;
-                       }
-               } else {
-                       if ((fr->frametype == AST_FRAME_DTMF_BEGIN) ||
-                           (fr->frametype == AST_FRAME_DTMF_END) ||
-                           (fr->frametype == AST_FRAME_VOICE) ||
-                           (fr->frametype == AST_FRAME_VIDEO) ||
-                           (fr->frametype == AST_FRAME_IMAGE) ||
-                           (fr->frametype == AST_FRAME_HTML) ||
-                           (fr->frametype == AST_FRAME_MODEM) ||
-                           (fr->frametype == AST_FRAME_TEXT)) {
-                               ast_write(other, fr);
-                       }
+       /* Lock both channels so we can look for the glue that binds them together */
+       ast_channel_lock_both(c_dst, c_src);
 
-                       ast_frfree(fr);
-               }
-               /* Swap priority */
-               cs[2] = cs[0];
-               cs[0] = cs[1];
-               cs[1] = cs[2];
+       if (!cap_src || !cap_dst) {
+               goto done;
        }
 
-       /* 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);
+       /* Grab glue that binds each channel to something using the RTP engine */
+       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;
        }
 
-       instance0->bridged = NULL;
-       instance1->bridged = NULL;
-
-       ast_poll_channel_del(c0, c1);
+       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;
 
-       return res;
-}
-
-static enum ast_bridge_result remote_bridge_loop(struct ast_channel *c0, struct ast_channel *c1, struct ast_rtp_instance *instance0, struct ast_rtp_instance *instance1,
-                                                struct ast_rtp_instance *vinstance0, struct ast_rtp_instance *vinstance1, struct ast_rtp_instance *tinstance0,
-                                                struct ast_rtp_instance *tinstance1, struct ast_rtp_glue *glue0, struct ast_rtp_glue *glue1, int codec0, int codec1, int timeoutms,
-                                                int flags, struct ast_frame **fo, struct ast_channel **rc, void *pvt0, void *pvt1)
-{
-       enum ast_bridge_result res = AST_BRIDGE_FAILED;
-       struct ast_channel *who = NULL, *other = NULL, *cs[3] = { NULL, };
-       int oldcodec0 = codec0, oldcodec1 = codec1;
-       struct sockaddr_in ac1 = {0,}, vac1 = {0,}, tac1 = {0,}, ac0 = {0,}, vac0 = {0,}, tac0 = {0,};
-       struct sockaddr_in t1 = {0,}, vt1 = {0,}, tt1 = {0,}, t0 = {0,}, vt0 = {0,}, tt0 = {0,};
-       struct ast_frame *fr = NULL;
+       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;
 
-       /* 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);
+       /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
+       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;
        }
-
-       /* Test the second channel */
-       if (!(glue1->update_peer(c1, instance0, vinstance0, tinstance0, codec0, 0))) {
-               ast_rtp_instance_get_remote_address(instance0, &ac0);
-               if (vinstance0) {
-                       ast_rtp_instance_get_remote_address(instance0, &vac0);
-               }
-               if (tinstance0) {
-                       ast_rtp_instance_get_remote_address(instance0, &tac0);
-               }
-       } else {
-               ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c1->name, c0->name);
+       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);
        }
 
-       ast_channel_unlock(c0);
-       ast_channel_unlock(c1);
-
-       instance0->bridged = instance1;
-       instance1->bridged = instance0;
-
-       ast_poll_channel_add(c0, c1);
-
-       /* Go into a loop handling any stray frames that may come in */
-       cs[0] = c0;
-       cs[1] = c1;
-       cs[2] = NULL;
-       for (;;) {
-               /* Check if anything changed */
-               if ((c0->tech_pvt != pvt0) ||
-                   (c1->tech_pvt != pvt1) ||
-                   (c0->masq || c0->masqr || c1->masq || c1->masqr) ||
-                   (c0->monitor || c0->audiohooks || c1->monitor || c1->audiohooks)) {
-                       ast_debug(1, "Oooh, something is weird, backing out\n");
-                       res = AST_BRIDGE_RETRY;
-                       break;
-               }
-
-               /* Check if they have changed their address */
-               ast_rtp_instance_get_remote_address(instance1, &t1);
-               if (vinstance1) {
-                       ast_rtp_instance_get_remote_address(vinstance1, &vt1);
-               }
-               if (tinstance1) {
-                       ast_rtp_instance_get_remote_address(tinstance1, &tt1);
-               }
-               if (glue1->get_codec) {
-                       codec1 = glue1->get_codec(c1);
-               }
+       /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
+       if (audio_glue_dst_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue_src_res != AST_RTP_GLUE_RESULT_REMOTE) {
+               goto done;
+       }
 
-               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);
-               }
+       /* Make sure we have matching codecs */
+       if (!ast_format_cap_has_joint(cap_dst, cap_src)) {
+               goto done;
+       }
 
-               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)) ||
-                   (codec0 != oldcodec0)) {
-                       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;
-               }
+       ast_rtp_codecs_payloads_copy(&instance_src->codecs, &instance_dst->codecs, instance_dst);
 
-               /* Wait for frame to come in on the channels */
-               if (!(who = ast_waitfor_n(cs, 2, &timeoutms))) {
-                       if (!timeoutms) {
-                               res = AST_BRIDGE_RETRY;
-                               break;
-                       }
-                       ast_debug(1, "Ooh, empty read...\n");
-                       if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
-                               break;
-                       }
-                       continue;
-               }
-               fr = ast_read(who);
-               other = (who == c0) ? c1 : c0;
-               if (!fr || ((fr->frametype == AST_FRAME_DTMF_BEGIN || fr->frametype == AST_FRAME_DTMF_END) &&
-                           (((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) ||
-                            ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1))))) {
-                       /* Break out of bridge */
-                       *fo = fr;
-                       *rc = who;
-                       ast_debug(1, "Oooh, got a %s\n", fr ? "digit" : "hangup");
-                       res = AST_BRIDGE_COMPLETE;
-                       break;
-               } else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
-                       if ((fr->subclass == AST_CONTROL_HOLD) ||
-                           (fr->subclass == AST_CONTROL_UNHOLD) ||
-                           (fr->subclass == AST_CONTROL_VIDUPDATE) ||
-                           (fr->subclass == AST_CONTROL_T38) ||
-                           (fr->subclass == AST_CONTROL_SRCUPDATE)) {
-                               if (fr->subclass == AST_CONTROL_HOLD) {
-                                       /* If we someone went on hold we want the other side to reinvite back to us */
-                                       if (who == c0) {
-                                               glue1->update_peer(c1, NULL, NULL, NULL, 0, 0);
-                                       } else {
-                                               glue0->update_peer(c0, NULL, NULL, NULL, 0, 0);
-                                       }
-                               } else if (fr->subclass == AST_CONTROL_UNHOLD) {
-                                       /* If they went off hold they should go back to being direct */
-                                       if (who == c0) {
-                                               glue1->update_peer(c1, instance0, vinstance0, tinstance0, codec0, 0);
-                                       } else {
-                                               glue0->update_peer(c0, instance1, vinstance1, tinstance1, codec1, 0);
-                                       }
-                               }
-                               /* Update local address information */
-                               ast_rtp_instance_get_remote_address(instance0, &t0);
-                               memcpy(&ac0, &t0, sizeof(ac0));
-                               ast_rtp_instance_get_remote_address(instance1, &t1);
-                               memcpy(&ac1, &t1, sizeof(ac1));
-                               /* Update codec information */
-                               if (glue0->get_codec && c0->tech_pvt) {
-                                       oldcodec0 = codec0 = glue0->get_codec(c0);
-                               }
-                               if (glue1->get_codec && c1->tech_pvt) {
-                                       oldcodec1 = codec1 = glue1->get_codec(c1);
-                               }
-                               ast_indicate_data(other, fr->subclass, fr->data.ptr, fr->datalen);
-                               ast_frfree(fr);
-                       } else {
-                               *fo = fr;
-                               *rc = who;
-                               ast_debug(1, "Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass, who->name);
-                               return AST_BRIDGE_COMPLETE;
-                       }
-               } else {
-                       if ((fr->frametype == AST_FRAME_DTMF_BEGIN) ||
-                           (fr->frametype == AST_FRAME_DTMF_END) ||
-                           (fr->frametype == AST_FRAME_VOICE) ||
-                           (fr->frametype == AST_FRAME_VIDEO) ||
-                           (fr->frametype == AST_FRAME_IMAGE) ||
-                           (fr->frametype == AST_FRAME_HTML) ||
-                           (fr->frametype == AST_FRAME_MODEM) ||
-                           (fr->frametype == AST_FRAME_TEXT)) {
-                               ast_write(other, fr);
-                       }
-                       ast_frfree(fr);
-               }
-               /* Swap priority */
-               cs[2] = cs[0];
-               cs[0] = cs[1];
-               cs[1] = cs[2];
+       if (vinstance_dst && vinstance_src) {
+               ast_rtp_codecs_payloads_copy(&vinstance_src->codecs, &vinstance_dst->codecs, vinstance_dst);
        }
-
-       if (glue0->update_peer(c0, NULL, NULL, NULL, 0, 0)) {
-               ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
+       if (tinstance_dst && tinstance_src) {
+               ast_rtp_codecs_payloads_copy(&tinstance_src->codecs, &tinstance_dst->codecs, tinstance_dst);
        }
-       if (glue1->update_peer(c1, NULL, NULL, NULL, 0, 0)) {
-               ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
+
+       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_debug(1, "Seeded SDP of '%s' with that of '%s'\n",
+                       ast_channel_name(c_dst), ast_channel_name(c_src));
        }
 
-       instance0->bridged = NULL;
-       instance1->bridged = NULL;
+done:
+       ast_channel_unlock(c_dst);
+       ast_channel_unlock(c_src);
 
-       ast_poll_channel_del(c0, c1);
+       ast_format_cap_destroy(cap_dst);
+       ast_format_cap_destroy(cap_src);
 
-       return res;
-}
-
-/*!
- * \brief Conditionally unref an rtp instance
- */
-static void unref_instance_cond(struct ast_rtp_instance **instance)
-{
-       if (*instance) {
-               ao2_ref(*instance, -1);
-               *instance = NULL;
-       }
-}
-
-enum ast_bridge_result ast_rtp_instance_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms)
-{
-       struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
-                       *vinstance0 = NULL, *vinstance1 = NULL,
-                       *tinstance0 = NULL, *tinstance1 = NULL;
-       struct ast_rtp_glue *glue0, *glue1;
-       enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID, text_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
-       enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID, text_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
-       enum ast_bridge_result res = AST_BRIDGE_FAILED;
-       int codec0 = 0, codec1 = 0;
-       int unlock_chans = 1;
-
-       /* Lock both channels so we can look for the glue that binds them together */
-       ast_channel_lock(c0);
-       while (ast_channel_trylock(c1)) {
-               ast_channel_unlock(c0);
-               usleep(1);
-               ast_channel_lock(c0);
-       }
-
-       /* Ensure neither channel got hungup during lock avoidance */
-       if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
-               ast_log(LOG_WARNING, "Got hangup while attempting to bridge '%s' and '%s'\n", c0->name, c1->name);
-               goto done;
-       }
-
-       /* Grab glue that binds each channel to something using the RTP engine */
-       if (!(glue0 = ast_rtp_instance_get_glue(c0->tech->type)) || !(glue1 = ast_rtp_instance_get_glue(c1->tech->type))) {
-               ast_debug(1, "Can't find native functions for channel '%s'\n", glue0 ? c1->name : c0->name);
-               goto done;
-       }
-
-       audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
-       video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
-       text_glue0_res = glue0->get_trtp_info ? glue0->get_trtp_info(c0, &tinstance0) : AST_RTP_GLUE_RESULT_FORBID;
-
-       audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
-       video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
-       text_glue1_res = glue1->get_trtp_info ? glue1->get_trtp_info(c1, &tinstance1) : AST_RTP_GLUE_RESULT_FORBID;
-
-       /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
-       if (video_glue0_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue0_res != AST_RTP_GLUE_RESULT_REMOTE)) {
-               audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
-       }
-       if (video_glue1_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue1_res != AST_RTP_GLUE_RESULT_REMOTE)) {
-               audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
-       }
-
-       /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
-       if (audio_glue0_res == AST_RTP_GLUE_RESULT_FORBID || audio_glue1_res == AST_RTP_GLUE_RESULT_FORBID) {
-               res = AST_BRIDGE_FAILED_NOWARN;
-               goto done;
-       }
-
-       /* If we need to get DTMF see if we can do it outside of the RTP stream itself */
-       if ((flags & AST_BRIDGE_DTMF_CHANNEL_0) && instance0->properties[AST_RTP_PROPERTY_DTMF]) {
-               res = AST_BRIDGE_FAILED_NOWARN;
-               goto done;
-       }
-       if ((flags & AST_BRIDGE_DTMF_CHANNEL_1) && instance1->properties[AST_RTP_PROPERTY_DTMF]) {
-               res = AST_BRIDGE_FAILED_NOWARN;
-               goto done;
-       }
-
-       /* If we have gotten to a local bridge make sure that both sides have the same local bridge callback and that they are DTMF compatible */
-       if ((audio_glue0_res == AST_RTP_GLUE_RESULT_LOCAL || audio_glue1_res == AST_RTP_GLUE_RESULT_LOCAL) && ((instance0->engine->local_bridge != instance1->engine->local_bridge) || (instance0->engine->dtmf_compatible && !instance0->engine->dtmf_compatible(c0, instance0, c1, instance1)))) {
-               res = AST_BRIDGE_FAILED_NOWARN;
-               goto done;
-       }
-
-       /* Make sure that codecs match */
-       codec0 = glue0->get_codec ? glue0->get_codec(c0) : 0;
-       codec1 = glue1->get_codec ? glue1->get_codec(c1) : 0;
-       if (codec0 && codec1 && !(codec0 & codec1)) {
-               ast_debug(1, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1);
-               res = AST_BRIDGE_FAILED_NOWARN;
-               goto done;
-       }
-
-       /* Depending on the end result for bridging either do a local bridge or remote bridge */
-       if (audio_glue0_res == AST_RTP_GLUE_RESULT_LOCAL || audio_glue1_res == AST_RTP_GLUE_RESULT_LOCAL) {
-               ast_verbose(VERBOSE_PREFIX_3 "Locally bridging %s and %s\n", c0->name, c1->name);
-               res = local_bridge_loop(c0, c1, instance0, instance1, timeoutms, flags, fo, rc, c0->tech_pvt, c1->tech_pvt);
-       } else {
-               ast_verbose(VERBOSE_PREFIX_3 "Remotely bridging %s and %s\n", c0->name, c1->name);
-               res = remote_bridge_loop(c0, c1, instance0, instance1, vinstance0, vinstance1,
-                               tinstance0, tinstance1, glue0, glue1, codec0, codec1, timeoutms, flags,
-                               fo, rc, c0->tech_pvt, c1->tech_pvt);
-       }
-
-       unlock_chans = 0;
-
-done:
-       if (unlock_chans) {
-               ast_channel_unlock(c0);
-               ast_channel_unlock(c1);
-       }
-
-       unref_instance_cond(&instance0);
-       unref_instance_cond(&instance1);
-       unref_instance_cond(&vinstance0);
-       unref_instance_cond(&vinstance1);
-       unref_instance_cond(&tinstance0);
-       unref_instance_cond(&tinstance1);
-
-       return res;
-}
-
-struct ast_rtp_instance *ast_rtp_instance_get_bridged(struct ast_rtp_instance *instance)
-{
-       return instance->bridged;
-}
-
-void ast_rtp_instance_early_bridge_make_compatible(struct ast_channel *c0, struct ast_channel *c1)
-{
-       struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
-               *vinstance0 = NULL, *vinstance1 = NULL,
-               *tinstance0 = NULL, *tinstance1 = NULL;
-       struct ast_rtp_glue *glue0, *glue1;
-       enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID, text_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
-       enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID, text_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
-       int codec0 = 0, codec1 = 0;
-       int res = 0;
-
-       /* Lock both channels so we can look for the glue that binds them together */
-       ast_channel_lock(c0);
-       while (ast_channel_trylock(c1)) {
-               ast_channel_unlock(c0);
-               usleep(1);
-               ast_channel_lock(c0);
-       }
-
-       /* Grab glue that binds each channel to something using the RTP engine */
-       if (!(glue0 = ast_rtp_instance_get_glue(c0->tech->type)) || !(glue1 = ast_rtp_instance_get_glue(c1->tech->type))) {
-               ast_debug(1, "Can't find native functions for channel '%s'\n", glue0 ? c1->name : c0->name);
-               goto done;
-       }
-
-       audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
-       video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
-       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>");
-       }
+       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);
 }
 
 int ast_rtp_instance_early_bridge(struct ast_channel *c0, struct ast_channel *c1)
@@ -1340,37 +1173,32 @@ int ast_rtp_instance_early_bridge(struct ast_channel *c0, struct ast_channel *c1
                        *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_NOLOCK);
+       struct ast_format_cap *cap1 = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK);
 
        /* If there is no second channel just immediately bail out, we are of no use in that scenario */
-       if (!c1) {
+       if (!c1 || !cap1 || !cap0) {
+               ast_format_cap_destroy(cap0);
+               ast_format_cap_destroy(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)) {
@@ -1379,11 +1207,11 @@ 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 */
@@ -1392,21 +1220,22 @@ 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_has_joint(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);
 
+       ast_format_cap_destroy(cap0);
+       ast_format_cap_destroy(cap1);
+
        unref_instance_cond(&instance0);
        unref_instance_cond(&instance1);
        unref_instance_cond(&vinstance0);
@@ -1414,11 +1243,9 @@ 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)
@@ -1461,8 +1288,8 @@ 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));
@@ -1478,44 +1305,72 @@ 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 = ast_channel_bridge_peer(chan);
+
+       ast_channel_lock(chan);
+       ast_channel_stage_snapshot(chan);
+       ast_channel_unlock(chan);
+       if (bridge) {
+               ast_channel_lock(bridge);
+               ast_channel_stage_snapshot(bridge);
+               ast_channel_unlock(bridge);
+       }
 
-       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_lock(chan);
+       ast_channel_stage_snapshot_done(chan);
+       ast_channel_unlock(chan);
+       if (bridge) {
+               ast_channel_lock(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;
 }
@@ -1532,17 +1387,21 @@ 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);
                ao2_ref(peer_instance, -1);
-               peer_instance = NULL;
                return -1;
        }
 
@@ -1556,12 +1415,26 @@ int ast_rtp_instance_make_compatible(struct ast_channel *chan, struct ast_rtp_in
        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_is_empty(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);
@@ -1578,6 +1451,11 @@ 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;
@@ -1587,3 +1465,606 @@ 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 res = 0;
+
+       if (!res_srtp) {
+               return -1;
+       }
+
+       if (!instance->srtp) {
+               res = res_srtp->create(&instance->srtp, instance, remote_policy);
+       } else {
+               res = res_srtp->replace(&instance->srtp, instance, remote_policy);
+       }
+       if (!res) {
+               res = res_srtp->add_stream(instance->srtp, local_policy);
+       }
+
+       return res;
+}
+
+struct ast_srtp *ast_rtp_instance_get_srtp(struct ast_rtp_instance *instance)
+{
+       return instance->srtp;
+}
+
+int ast_rtp_instance_sendcng(struct ast_rtp_instance *instance, int level)
+{
+       if (instance->engine->sendcng) {
+               return instance->engine->sendcng(instance, level);
+       }
+
+       return -1;
+}
+
+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")) {
+               dtls_cfg->verify = ast_true(value) ? 1 : 0;
+       } 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);
+               dtls_cfg->certfile = ast_strdup(value);
+       } else if (!strcasecmp(name, "dtlsprivatekey")) {
+               ast_free(dtls_cfg->pvtfile);
+               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);
+               dtls_cfg->cafile = ast_strdup(value);
+       } else if (!strcasecmp(name, "dtlscapath") || !strcasecmp(name, "dtlscadir")) {
+               ast_free(dtls_cfg->capath);
+               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 {
+               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)
+{
+       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->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);
+       ast_free(dtls_cfg->pvtfile);
+       ast_free(dtls_cfg->cipher);
+       ast_free(dtls_cfg->cafile);
+       ast_free(dtls_cfg->capath);
+}
+
+static void set_next_mime_type(const struct ast_format *format, int rtp_code, char *type, char *subtype, unsigned int sample_rate)
+{
+       int x = mime_types_len;
+       if (ARRAY_LEN(ast_rtp_mime_types) == mime_types_len) {
+               return;
+       }
+
+       ast_rwlock_wrlock(&mime_types_lock);
+       if (format) {
+               ast_rtp_mime_types[x].payload_type.asterisk_format = 1;
+               ast_format_copy(&ast_rtp_mime_types[x].payload_type.format, format);
+       } else {
+               ast_rtp_mime_types[x].payload_type.rtp_code = rtp_code;
+       }
+       ast_rtp_mime_types[x].type = type;
+       ast_rtp_mime_types[x].subtype = 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, const struct ast_format *format, int rtp_code)
+{
+       int x;
+       ast_rwlock_wrlock(&static_RTP_PT_lock);
+       if (map < 0) {
+               /* find next available dynamic payload slot */
+               for (x = 96; x < 127; x++) {
+                       if (!static_RTP_PT[x].asterisk_format && !static_RTP_PT[x].rtp_code) {
+                               map = x;
+                               break;
+                       }
+               }
+       }
+
+       if (map < 0) {
+               ast_log(LOG_WARNING, "No Dynamic RTP mapping available for format %s\n" ,ast_getformatname(format));
+               ast_rwlock_unlock(&static_RTP_PT_lock);
+               return;
+       }
+
+       if (format) {
+               static_RTP_PT[map].asterisk_format = 1;
+               ast_format_copy(&static_RTP_PT[map].format, format);
+       } else {
+               static_RTP_PT[map].rtp_code = rtp_code;
+       }
+       ast_rwlock_unlock(&static_RTP_PT_lock);
+}
+
+int ast_rtp_engine_load_format(const struct ast_format *format)
+{
+       switch (format->id) {
+       case AST_FORMAT_SILK:
+               set_next_mime_type(format, 0, "audio", "SILK", ast_format_rate(format));
+               add_static_payload(-1, format, 0);
+               break;
+       case AST_FORMAT_CELT:
+               set_next_mime_type(format, 0, "audio", "CELT", ast_format_rate(format));
+               add_static_payload(-1, format, 0);
+               break;
+       default:
+               break;
+       }
+
+       return 0;
+}
+
+int ast_rtp_engine_unload_format(const 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 (ast_format_cmp(&static_RTP_PT[x].format, format) == AST_FORMAT_CMP_EQUAL) {
+                       memset(&static_RTP_PT[x], 0, sizeof(struct ast_rtp_payload_type));
+               }
+       }
+       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) {
+                       continue;
+               }
+               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 * 4096);
+               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; i++) {
+               struct ast_json *json_report_block;
+               json_report_block = ast_json_pack("{s: i, s: i, s: i, s: i, s: i, s: i, 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", payload->report->report_block[i]->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) {
+               json_rtcp_sender_info = ast_json_pack("{s: i, s: i, s: i, s: i, s: i}",
+                               "ntp_timestamp_sec", payload->report->sender_information.ntp_timestamp.tv_sec,
+                               "ntp_timestamp_usec", payload->report->sender_information.ntp_timestamp.tv_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 ? json_rtcp_sender_info : ast_json_null(),
+                       "report_blocks", 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 ? json_channel : ast_json_null(),
+               "rtcp_report", json_rtcp_report,
+               "blob", 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);
+
+       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)
+{
+       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);
+}
+
+int ast_rtp_engine_init()
+{
+       struct ast_format tmpfmt;
+
+       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_atexit(rtp_engine_shutdown);
+
+       /* Define all the RTP mime types available */
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G723_1, 0), 0, "audio", "G723", 8000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_GSM, 0), 0, "audio", "GSM", 8000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0), 0, "audio", "PCMU", 8000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0), 0, "audio", "G711U", 8000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ALAW, 0), 0, "audio", "PCMA", 8000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ALAW, 0), 0, "audio", "G711A", 8000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G726, 0), 0, "audio", "G726-32", 8000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0, "audio", "DVI4", 8000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0), 0, "audio", "L16", 8000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR16, 0), 0, "audio", "L16", 16000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR16, 0), 0, "audio", "L16-256", 16000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_LPC10, 0), 0, "audio", "LPC", 8000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0, "audio", "G729", 8000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0, "audio", "G729A", 8000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0, "audio", "G.729", 8000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SPEEX, 0), 0, "audio", "speex", 8000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SPEEX16, 0), 0,  "audio", "speex", 16000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SPEEX32, 0), 0,  "audio", "speex", 32000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ILBC, 0), 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_set(&tmpfmt, AST_FORMAT_G722, 0), 0, "audio", "G722", 8000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G726_AAL2, 0), 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_set(&tmpfmt, AST_FORMAT_JPEG, 0), 0, "video", "JPEG", 90000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_PNG, 0), 0, "video", "PNG", 90000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H261, 0), 0, "video", "H261", 90000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H263, 0), 0, "video", "H263", 90000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H263_PLUS, 0), 0, "video", "H263-1998", 90000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H264, 0), 0, "video", "H264", 90000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_MP4_VIDEO, 0), 0, "video", "MP4V-ES", 90000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_T140RED, 0), 0, "text", "RED", 1000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_T140, 0), 0, "text", "T140", 1000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SIREN7, 0), 0, "audio", "G7221", 16000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SIREN14, 0), 0, "audio", "G7221", 32000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G719, 0), 0, "audio", "G719", 48000);
+       /* Opus and VP8 */
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_OPUS, 0), 0,  "audio", "opus", 48000);
+       set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_VP8, 0), 0,  "video", "VP8", 90000);
+
+       /* Define the static rtp payload mappings */
+       add_static_payload(0, ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0), 0);
+       #ifdef USE_DEPRECATED_G726
+       add_static_payload(2, ast_format_set(&tmpfmt, AST_FORMAT_G726, 0), 0);/* Technically this is G.721, but if Cisco can do it, so can we... */
+       #endif
+       add_static_payload(3, ast_format_set(&tmpfmt, AST_FORMAT_GSM, 0), 0);
+       add_static_payload(4, ast_format_set(&tmpfmt, AST_FORMAT_G723_1, 0), 0);
+       add_static_payload(5, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0);/* 8 kHz */
+       add_static_payload(6, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0); /* 16 kHz */
+       add_static_payload(7, ast_format_set(&tmpfmt, AST_FORMAT_LPC10, 0), 0);
+       add_static_payload(8, ast_format_set(&tmpfmt, AST_FORMAT_ALAW, 0), 0);
+       add_static_payload(9, ast_format_set(&tmpfmt, AST_FORMAT_G722, 0), 0);
+       add_static_payload(10, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0), 0); /* 2 channels */
+       add_static_payload(11, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0), 0); /* 1 channel */
+       add_static_payload(13, NULL, AST_RTP_CN);
+       add_static_payload(16, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0); /* 11.025 kHz */
+       add_static_payload(17, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0); /* 22.050 kHz */
+       add_static_payload(18, ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0);
+       add_static_payload(19, NULL, AST_RTP_CN);         /* Also used for CN */
+       add_static_payload(26, ast_format_set(&tmpfmt, AST_FORMAT_JPEG, 0), 0);
+       add_static_payload(31, ast_format_set(&tmpfmt, AST_FORMAT_H261, 0), 0);
+       add_static_payload(34, ast_format_set(&tmpfmt, AST_FORMAT_H263, 0), 0);
+       add_static_payload(97, ast_format_set(&tmpfmt, AST_FORMAT_ILBC, 0), 0);
+       add_static_payload(98, ast_format_set(&tmpfmt, AST_FORMAT_H263_PLUS, 0), 0);
+       add_static_payload(99, ast_format_set(&tmpfmt, AST_FORMAT_H264, 0), 0);
+       add_static_payload(101, NULL, AST_RTP_DTMF);
+       add_static_payload(102, ast_format_set(&tmpfmt, AST_FORMAT_SIREN7, 0), 0);
+       add_static_payload(103, ast_format_set(&tmpfmt, AST_FORMAT_H263_PLUS, 0), 0);
+       add_static_payload(104, ast_format_set(&tmpfmt, AST_FORMAT_MP4_VIDEO, 0), 0);
+       add_static_payload(105, ast_format_set(&tmpfmt, AST_FORMAT_T140RED, 0), 0);   /* Real time text chat (with redundancy encoding) */
+       add_static_payload(106, ast_format_set(&tmpfmt, AST_FORMAT_T140, 0), 0);     /* Real time text chat */
+       add_static_payload(110, ast_format_set(&tmpfmt, AST_FORMAT_SPEEX, 0), 0);
+       add_static_payload(111, ast_format_set(&tmpfmt, AST_FORMAT_G726, 0), 0);
+       add_static_payload(112, ast_format_set(&tmpfmt, AST_FORMAT_G726_AAL2, 0), 0);
+       add_static_payload(115, ast_format_set(&tmpfmt, AST_FORMAT_SIREN14, 0), 0);
+       add_static_payload(116, ast_format_set(&tmpfmt, AST_FORMAT_G719, 0), 0);
+       add_static_payload(117, ast_format_set(&tmpfmt, AST_FORMAT_SPEEX16, 0), 0);
+       add_static_payload(118, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR16, 0), 0); /* 16 Khz signed linear */
+       add_static_payload(119, ast_format_set(&tmpfmt, AST_FORMAT_SPEEX32, 0), 0);
+       add_static_payload(121, NULL, AST_RTP_CISCO_DTMF);   /* Must be type 121 */
+       /* Opus and VP8 */
+       add_static_payload(100, ast_format_set(&tmpfmt, AST_FORMAT_VP8, 0), 0);
+       add_static_payload(107, ast_format_set(&tmpfmt, AST_FORMAT_OPUS, 0), 0);
+
+       return 0;
+}