diff --git a/build_tools/menuselect-deps.in b/build_tools/menuselect-deps.in
index 6594a922eb..1d9703419a 100644
--- a/build_tools/menuselect-deps.in
+++ b/build_tools/menuselect-deps.in
@@ -57,6 +57,7 @@ SPANDSP=@PBX_SPANDSP@
 SPEEX=@PBX_SPEEX@
 SPEEXDSP=@PBX_SPEEXDSP@
 SPEEX_PREPROCESS=@PBX_SPEEX_PREPROCESS@
+VEVS=@PBX_VEVS@
 SQLITE3=@PBX_SQLITE3@
 SRTP=@PBX_SRTP@
 SS7=@PBX_SS7@
diff --git a/codecs/codec_vevs.c b/codecs/codec_vevs.c
new file mode 100644
index 0000000000..14de95c4dd
--- /dev/null
+++ b/codecs/codec_vevs.c
@@ -0,0 +1,285 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ */
+
+/*! \file
+ *
+ * \brief Translate between signed linear and Vocal EVS codec
+ *
+ * \ingroup codecs
+ */
+
+/*** MODULEINFO
+	<depend>vevs</depend>
+ ***/
+
+#include "asterisk.h"
+
+#include "asterisk/translate.h"
+#include "asterisk/config.h"
+#include "asterisk/module.h"
+#include "asterisk/utils.h"
+#include "asterisk/linkedlists.h"
+
+#include <stdbool.h>
+#define uint8 uint8_t
+#define uint16 uint16_t
+#define uint32 uint32_t
+#define vbool bool
+#define sint15 int16_t
+#include "vocal-evs/evs.h"
+
+#define BUFFER_SAMPLES	8000		/* 0.5 seconds @ 16 kHz */
+#define BUFFER_FRAMES	8192		/* Enough for about 0.5 seconds @ 128 Kbps */
+
+/* Sample frame data */
+
+#include "asterisk/slin.h"
+#include "ex_vevs.h"
+
+/* Encoder and decoder instances */
+
+struct evs_enc_pvt {
+	evs_enc_ctx_t enc;		/* Encoder states */
+	int chunk;			/* Size of chunk to encode (in samples) */
+	int16_t buf[BUFFER_SAMPLES];	/* Buffer to store received uncompressed audio */
+};
+
+struct evs_dec_pvt {
+	evs_dec_ctx_t dec;		/* Decoder states */
+	int chunk;			/* Size of chunk to decode (in samples) */
+};
+
+#warning hacking
+int vocal_evs_init_coder (evs_enc_ctx_t *ctx, uint16 bandwidth, uint16 sampling_frequency, uint32 bitrate, vbool dtx_enable) { return 320; }
+int vocal_evs_process_coder (evs_enc_ctx_t *ctx, uint8 *enc_data, uint16 enc_data_size, const sint15 *enc_samples, uint16 sample_count) { return 320; }
+int vocal_evs_close_coder (evs_enc_ctx_t *ctx) { return 0; }
+int vocal_evs_init_decoder (evs_dec_ctx_t *ctx, uint16 sampling_frequency) { return 320; }
+int vocal_evs_process_decoder (evs_dec_ctx_t *ctx, sint15 *dec_samples, uint16 max_sample_count, uint8 *dec_data, uint16 dec_data_size) { return 320; }
+int vocal_evs_close_decoder (evs_dec_ctx_t *ctx) { return 0; }
+
+
+/* Create and destroy codec intances */
+
+static int evs_enc_new(struct ast_trans_pvt *pvt)
+{
+	struct evs_enc_pvt *enc = pvt->pvt;
+	const unsigned int sample_rate = pvt->t->src_codec.sample_rate;
+	int rc;
+
+	memset(enc, 0, sizeof(*enc));
+	rc = vocal_evs_init_coder(&enc->enc, EVS_BW_NB, EVS_SF_16K, EVS_BR_5K90, false);
+	if (rc <= 0) {
+		ast_log(LOG_ERROR, "Error creating the Vocal EVS encoder\n");
+		return -1;
+	}
+	enc->chunk = rc;
+
+	ast_debug(3, "Created encoder (Vocal EVS) with sample rate %d\n", sample_rate);
+
+	return 0;
+}
+
+static void evs_enc_destroy(struct ast_trans_pvt *pvt)
+{
+	struct evs_enc_pvt *enc = pvt->pvt;
+
+	vocal_evs_close_coder(&enc->enc);
+
+	ast_debug(3, "Destroyed encoder (Vocal EVS)\n");
+}
+
+static int evs_dec_new(struct ast_trans_pvt *pvt)
+{
+	struct evs_dec_pvt *dec = pvt->pvt;
+	const unsigned int sample_rate = pvt->t->src_codec.sample_rate;
+	int rc;
+
+	memset(dec, 0, sizeof(*dec));
+	rc = vocal_evs_init_decoder(&dec->dec, EVS_SF_16K);
+	if (rc <= 0) {
+		ast_log(LOG_ERROR, "Error creating the Vocal EVS decoder\n");
+		return -1;
+	}
+	dec->chunk = rc;
+
+	ast_debug(3, "Created decoder (Vocal EVS) with sample rate %d\n", sample_rate);
+
+	return 0;
+}
+
+static void evs_dec_destroy(struct ast_trans_pvt *pvt)
+{
+	struct evs_dec_pvt *dec = pvt->pvt;
+
+	vocal_evs_close_decoder(&dec->dec);
+
+	ast_debug(3, "Destroyed encoder (Vocal EVS)\n");
+}
+
+/* Encoder */
+
+static int lintoevs_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
+{
+	struct evs_enc_pvt *enc = pvt->pvt;
+
+	/* XXX We should look at how old the rest of our stream is, and if it
+	   is too old, then we should overwrite it entirely, otherwise we can
+	   get artifacts of earlier talk that do not belong */
+	if (pvt->samples + f->samples > BUFFER_SAMPLES) {
+		ast_log(LOG_WARNING, "Out of buffer space\n");
+		return -1;
+	}
+	memcpy(enc->buf + pvt->samples, f->data.ptr, f->datalen);
+	pvt->samples += f->samples;
+	return 0;
+}
+
+static struct ast_frame *lintoevs_frameout(struct ast_trans_pvt *pvt)
+{
+	struct evs_enc_pvt *enc = pvt->pvt;
+	struct ast_frame *result = NULL;
+	struct ast_frame *last = NULL;
+	int samples = 0; /* output samples */
+
+	while (pvt->samples >= enc->chunk) {
+		struct ast_frame *current;
+		int rc;
+
+		rc = vocal_evs_process_coder(&enc->enc, pvt->outbuf.uc, BUFFER_FRAMES, enc->buf, enc->chunk);
+		if (rc <= 0) {
+			ast_log(LOG_ERROR, "Error encoding with Vocal EVS encoder\n");
+			pvt->samples = 0;
+			return NULL;
+		}
+		samples += enc->chunk;
+		pvt->samples -= enc->chunk;
+		current = ast_trans_frameout(pvt, rc, enc->chunk);
+		if (!current) {
+			continue;
+		} else if (last) {
+			/* Append frame */
+			AST_LIST_NEXT(last, frame_list) = current;
+		} else {
+			/* Return first frame */
+			result = current;
+		}
+		last = current;
+	}
+
+	/* Move the data at the end of the buffer to the front */
+	if (samples) {
+		memmove(enc->buf, enc->buf + samples, pvt->samples * sizeof(int16_t));
+	}
+
+	return result;
+}
+
+/* Decoder */
+
+static int evstolin_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
+{
+	struct evs_dec_pvt *dec = pvt->pvt;
+	int16_t *dst = pvt->outbuf.i16;
+	int rc;
+
+	if (BUFFER_SAMPLES - pvt->samples < dec->chunk) {
+		ast_log(LOG_WARNING, "Out of buffer space\n");
+		return -1;
+	}
+
+	if (f->datalen == 0) {
+		ast_log(LOG_DEBUG, "Conceal missing EVS frame.\n");
+		/* The documentation does not give a hint on how bad frames indicated and if they are concealed at all. This may not work!*/
+		rc = vocal_evs_process_decoder(&dec->dec, dst + pvt->samples, BUFFER_SAMPLES - pvt->samples, NULL, 0);
+		if (rc <= 0) {
+			ast_log(LOG_ERROR, "Error concealing missing EVS frame with Vocal EVS decoder\n");
+			return -1;
+		}
+	} else {
+		rc = vocal_evs_process_decoder(&dec->dec, dst + pvt->samples, BUFFER_SAMPLES - pvt->samples, f->data.ptr, f->datalen);
+		if (rc <= 0) {
+			ast_log(LOG_ERROR, "Error decoding with Vocal EVS decoder\n");
+			return -1;
+		}
+	}
+	pvt->samples += rc;
+	pvt->datalen += rc * sizeof(int16_t);
+
+	return 0;
+}
+
+static struct ast_translator lintoevs = {
+	.name = "lintovevs",
+	.src_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 16000,
+	},
+	.dst_codec = {
+		.name = "vevs",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 16000,
+	},
+	.format = "vevs",
+	.newpvt = evs_enc_new,
+	.framein = lintoevs_framein,
+	.frameout = lintoevs_frameout,
+	.destroy = evs_enc_destroy,
+	.sample = slin16_sample,
+	.buf_size = BUFFER_FRAMES,
+	.desc_size = sizeof (struct evs_enc_pvt ),
+};
+
+static struct ast_translator evstolin = {
+	.name = "vevstolin",
+	.src_codec = {
+		.name = "vevs",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 16000,
+	},
+	.dst_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 16000,
+	},
+	.format = "slin",
+	.newpvt = evs_dec_new,
+	.framein = evstolin_framein,
+	.destroy = evs_dec_destroy,
+	.sample = vevs_sample,
+	.buffer_samples = BUFFER_SAMPLES,
+	.buf_size = BUFFER_SAMPLES * 2,
+	.desc_size = sizeof (struct evs_dec_pvt ),
+};
+
+static int unload_module(void)
+{
+	int res;
+
+	res = ast_unregister_translator(&lintoevs);
+	res |= ast_unregister_translator(&evstolin);
+
+	return res;
+}
+
+static int load_module(void)
+{
+	int res;
+
+	res = ast_register_translator(&evstolin);
+	res |= ast_register_translator(&lintoevs);
+
+	if (res) {
+		unload_module();
+		return AST_MODULE_LOAD_DECLINE;
+	}
+
+	return AST_MODULE_LOAD_SUCCESS;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Vocal EVS Coder/Decoder",
+	.support_level = AST_MODULE_SUPPORT_CORE,
+	.load = load_module,
+	.unload = unload_module,
+);
diff --git a/codecs/ex_vevs.h b/codecs/ex_vevs.h
new file mode 100644
index 0000000000..84bbf5f57c
--- /dev/null
+++ b/codecs/ex_vevs.h
@@ -0,0 +1,28 @@
+#include "asterisk/format_cache.h"      /* for ast_format_evs */
+#include "asterisk/frame.h"             /* for ast_frame, etc */
+
+static uint8_t ex_vevs[] = { /* PRIMARY_16400, WB */
+	0x05, 0x51, 0x63, 0x4e, 0xa7, 0x87, 0x0c, 0xc4,
+	0x50, 0x3c, 0xcf, 0x60, 0x19, 0xbf, 0xd3, 0x93,
+	0xf4, 0xd9, 0x49, 0xac, 0x89, 0xce, 0x4c, 0x4d,
+	0x5e, 0x01, 0xff, 0x80, 0x00, 0x17, 0x17, 0xd5,
+	0x73, 0x7b, 0xd5, 0x1d, 0xe1, 0xcf, 0x65, 0x0b,
+	0xee, 0x95
+};
+
+static struct ast_frame *vevs_sample(void)
+{
+	static struct ast_frame f = {
+		.frametype = AST_FRAME_VOICE,
+		.datalen = sizeof(ex_vevs),
+		.samples = 320,
+		.mallocd = 0,
+		.offset = 0,
+		.src = __PRETTY_FUNCTION__,
+		.data.ptr = ex_vevs,
+	};
+
+	f.subclass.format = ast_format_vevs;
+
+	return &f;
+}
diff --git a/configure b/configure
index 07764740dc..a371b91c16 100755
--- a/configure
+++ b/configure
@@ -1017,6 +1017,10 @@ PBX_PJSIP_DLG_CREATE_UAS_AND_INC_LOCK
 PJSIP_DLG_CREATE_UAS_AND_INC_LOCK_DIR
 PJSIP_DLG_CREATE_UAS_AND_INC_LOCK_INCLUDE
 PJSIP_DLG_CREATE_UAS_AND_INC_LOCK_LIB
+PBX_VEVS
+VEVS_DIR
+VEVS_INCLUDE
+VEVS_LIB
 PBX_BEANSTALK
 BEANSTALK_DIR
 BEANSTALK_INCLUDE
@@ -2216,6 +2220,7 @@ Optional Packages:
   --with-opusfile=PATH    use Opusfile files in PATH
   --with-postgres=PATH    use PostgreSQL files in PATH
   --with-beanstalk=PATH   use Beanstalk Job Queue files in PATH
+  --with-bluetooth=PATH   use Vocal EVS Audio Decoder/Encoder files in PATH
   --with-pjproject=PATH   use PJPROJECT files in PATH
   --with-popt=PATH        use popt files in PATH
   --with-portaudio=PATH   use PortAudio files in PATH
@@ -13151,6 +13156,41 @@ fi
 
 
 
+#FIXME
+
+    VEVS_DESCRIP="Vocal EVS Audio Decoder/Encoder"
+    VEVS_OPTION="bluetooth"
+    PBX_VEVS=0
+
+# Check whether --with-bluetooth was given.
+if test ${with_bluetooth+y}
+then :
+  withval=$with_bluetooth;
+	case ${withval} in
+	n|no)
+	USE_VEVS=no
+	# -1 is a magic value used by menuselect to know that the package
+	# was disabled, other than 'not found'
+	PBX_VEVS=-1
+	;;
+	y|ye|yes)
+	ac_mandatory_list="${ac_mandatory_list} VEVS"
+	;;
+	*)
+	VEVS_DIR="${withval}"
+	ac_mandatory_list="${ac_mandatory_list} VEVS"
+	;;
+	esac
+
+fi
+
+
+
+
+
+
+
+#AST_EXT_LIB_SETUP([VEVS], [Vocal EVS Audio Decoder/Encoder], [vocal-evs])
 
 if test "x${PBX_PJPROJECT}" != "x1" ; then
 
@@ -26471,6 +26511,105 @@ fi
 
 
 
+#FIXME
+
+if test "x${PBX_VEVS}" != "x1" -a "${USE_VEVS}" != "no"; then
+   pbxlibdir=""
+   # if --with-VEVS=DIR has been specified, use it.
+   if test "x${VEVS_DIR}" != "x"; then
+      if test -d ${VEVS_DIR}/lib; then
+         pbxlibdir="-L${VEVS_DIR}/lib"
+      else
+         pbxlibdir="-L${VEVS_DIR}"
+      fi
+   fi
+
+      ast_ext_lib_check_save_CFLAGS="${CFLAGS}"
+      CFLAGS="${CFLAGS} "
+      { printf "%s\n" "$as_me:${as_lineno-$LINENO}: checking for ba2str in -lbluetooth" >&5
+printf %s "checking for ba2str in -lbluetooth... " >&6; }
+if test ${ac_cv_lib_bluetooth_ba2str+y}
+then :
+  printf %s "(cached) " >&6
+else $as_nop
+  ac_check_lib_save_LIBS=$LIBS
+LIBS="-lbluetooth ${pbxlibdir}  $LIBS"
+cat confdefs.h - <<_ACEOF >conftest.$ac_ext
+/* end confdefs.h.  */
+
+/* Override any GCC internal prototype to avoid an error.
+   Use char because int might match the return type of a GCC
+   builtin and then its argument prototype would still apply.  */
+char ba2str ();
+int
+main (void)
+{
+return ba2str ();
+  ;
+  return 0;
+}
+_ACEOF
+if ac_fn_c_try_link "$LINENO"
+then :
+  ac_cv_lib_bluetooth_ba2str=yes
+else $as_nop
+  ac_cv_lib_bluetooth_ba2str=no
+fi
+rm -f core conftest.err conftest.$ac_objext conftest.beam \
+    conftest$ac_exeext conftest.$ac_ext
+LIBS=$ac_check_lib_save_LIBS
+fi
+{ printf "%s\n" "$as_me:${as_lineno-$LINENO}: result: $ac_cv_lib_bluetooth_ba2str" >&5
+printf "%s\n" "$ac_cv_lib_bluetooth_ba2str" >&6; }
+if test "x$ac_cv_lib_bluetooth_ba2str" = xyes
+then :
+  AST_VEVS_FOUND=yes
+else $as_nop
+  AST_VEVS_FOUND=no
+fi
+
+      CFLAGS="${ast_ext_lib_check_save_CFLAGS}"
+
+
+   # now check for the header.
+   if test "${AST_VEVS_FOUND}" = "yes"; then
+      VEVS_LIB="${pbxlibdir} -lbluetooth "
+      # if --with-VEVS=DIR has been specified, use it.
+      if test "x${VEVS_DIR}" != "x"; then
+         VEVS_INCLUDE="-I${VEVS_DIR}/include"
+      fi
+      VEVS_INCLUDE="${VEVS_INCLUDE} "
+
+         # check for the header
+         ast_ext_lib_check_saved_CPPFLAGS="${CPPFLAGS}"
+         CPPFLAGS="${CPPFLAGS} ${VEVS_INCLUDE}"
+         ac_fn_c_check_header_compile "$LINENO" "bluetooth/bluetooth.h" "ac_cv_header_bluetooth_bluetooth_h" "$ac_includes_default"
+if test "x$ac_cv_header_bluetooth_bluetooth_h" = xyes
+then :
+  VEVS_HEADER_FOUND=1
+else $as_nop
+  VEVS_HEADER_FOUND=0
+fi
+
+         CPPFLAGS="${ast_ext_lib_check_saved_CPPFLAGS}"
+
+      if test "x${VEVS_HEADER_FOUND}" = "x0" ; then
+         VEVS_LIB=""
+         VEVS_INCLUDE=""
+      else
+
+         PBX_VEVS=1
+         cat >>confdefs.h <<_ACEOF
+#define HAVE_VEVS 1
+_ACEOF
+
+      fi
+   fi
+fi
+
+
+#AST_EXT_LIB_CHECK([VEVS], [vocal-evs], [vevs_enc], [vocal-evs/evs.h])
+
 PG_CONFIG=":"
 if test "${USE_PGSQL}" != "no"; then
    if test "x${PGSQL_DIR}" != "x"; then
diff --git a/configure.ac b/configure.ac
index 0e1a11eeda..ec5f5b1253 100644
--- a/configure.ac
+++ b/configure.ac
@@ -579,6 +579,9 @@ AST_EXT_LIB_SETUP([OPUS], [Opus], [opus])
 AST_EXT_LIB_SETUP([OPUSFILE], [Opusfile], [opusfile])
 AST_EXT_LIB_SETUP([PGSQL], [PostgreSQL], [postgres])
 AST_EXT_LIB_SETUP([BEANSTALK], [Beanstalk Job Queue], [beanstalk])
+#FIXME
+AST_EXT_LIB_SETUP([VEVS], [Vocal EVS Audio Decoder/Encoder], [bluetooth])
+#AST_EXT_LIB_SETUP([VEVS], [Vocal EVS Audio Decoder/Encoder], [vocal-evs])
 
 if test "x${PBX_PJPROJECT}" != "x1" ; then
 AST_EXT_LIB_SETUP([PJPROJECT], [PJPROJECT], [pjproject])
@@ -2409,6 +2412,10 @@ AST_EXT_LIB_CHECK([BLUETOOTH], [bluetooth], [ba2str], [bluetooth/bluetooth.h])
 
 AST_EXT_LIB_CHECK([BEANSTALK], [beanstalk], [bs_version], [beanstalk.h])
 
+#FIXME
+AST_EXT_LIB_CHECK([VEVS], [bluetooth], [ba2str], [bluetooth/bluetooth.h])
+#AST_EXT_LIB_CHECK([VEVS], [vocal-evs], [vevs_enc], [vocal-evs/evs.h])
+
 PG_CONFIG=":"
 if test "${USE_PGSQL}" != "no"; then
    if test "x${PGSQL_DIR}" != "x"; then
diff --git a/include/asterisk/autoconfig.h.in b/include/asterisk/autoconfig.h.in
index debf540c3b..023915f7d3 100644
--- a/include/asterisk/autoconfig.h.in
+++ b/include/asterisk/autoconfig.h.in
@@ -1226,6 +1226,9 @@
 /* Define to 1 if you have the `vasprintf' function. */
 #undef HAVE_VASPRINTF
 
+/* Define to 1 if you have the Vocal EVS Audio Decoder/Encoder library. */
+#undef HAVE_VEVS
+
 /* Define to 1 if you have the `vfork' function. */
 #undef HAVE_VFORK
 
diff --git a/include/asterisk/format_cache.h b/include/asterisk/format_cache.h
index 20b9d3afbb..c1c3689d15 100644
--- a/include/asterisk/format_cache.h
+++ b/include/asterisk/format_cache.h
@@ -245,6 +245,10 @@ extern struct ast_format *ast_format_silk8;
 extern struct ast_format *ast_format_silk12;
 extern struct ast_format *ast_format_silk16;
 extern struct ast_format *ast_format_silk24;
+/*!
+ * \brief Built-in cached Vocal EVS format.
+ */
+extern struct ast_format *ast_format_vevs;
 
 /*!
  * \brief Initialize format cache support within the core.
diff --git a/main/codec_builtin.c b/main/codec_builtin.c
index 02c9f2e79a..25591d3578 100644
--- a/main/codec_builtin.c
+++ b/main/codec_builtin.c
@@ -925,6 +925,16 @@ static struct ast_codec silk24 = {
 	.samples_count = silk_samples
 };
 
+static struct ast_codec vevs = {
+	.name = "vevs",
+	.description = "Vocal EVS",
+	.type = AST_MEDIA_TYPE_AUDIO,
+	.sample_rate = 16000,
+	.minimum_ms = 20,
+	.maximum_ms = 300,
+	.default_ms = 20,
+};
+
 #define CODEC_REGISTER_AND_CACHE(codec) \
 	({ \
 		int __res_ ## __LINE__ = 0; \
@@ -1003,6 +1013,7 @@ int ast_codec_builtin_init(void)
 	res |= CODEC_REGISTER_AND_CACHE_NAMED("silk12", silk12);
 	res |= CODEC_REGISTER_AND_CACHE_NAMED("silk16", silk16);
 	res |= CODEC_REGISTER_AND_CACHE_NAMED("silk24", silk24);
+	res |= CODEC_REGISTER_AND_CACHE(vevs);
 
 	return res;
 }
diff --git a/main/format_cache.c b/main/format_cache.c
index 74aa0fe093..0ad2cdeaef 100644
--- a/main/format_cache.c
+++ b/main/format_cache.c
@@ -252,6 +252,10 @@ struct ast_format *ast_format_silk8;
 struct ast_format *ast_format_silk12;
 struct ast_format *ast_format_silk16;
 struct ast_format *ast_format_silk24;
+/*!
+ * \brief Built-in cached Vocal EVS format.
+ */
+struct ast_format *ast_format_vevs;
 
 /*! \brief Number of buckets to use for the media format cache (should be prime for performance reasons) */
 #define CACHE_BUCKETS 53
@@ -359,6 +363,7 @@ static void format_cache_shutdown(void)
 	ao2_replace(ast_format_silk12, NULL);
 	ao2_replace(ast_format_silk16, NULL);
 	ao2_replace(ast_format_silk24, NULL);
+	ao2_replace(ast_format_vevs, NULL);
 }
 
 int ast_format_cache_init(void)
@@ -468,6 +473,8 @@ static void set_cached_format(const char *name, struct ast_format *format)
 		ao2_replace(ast_format_silk16, format);
 	} else if (!strcmp(name, "silk24")) {
 		ao2_replace(ast_format_silk24, format);
+	} else if (!strcmp(name, "vevs")) {
+		ao2_replace(ast_format_vevs, format);
 	}
 }
 
diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index 7009ad73fa..7ce5c70f43 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -3724,6 +3724,7 @@ int ast_rtp_engine_init(void)
 	set_next_mime_type(ast_format_opus, 0,  "audio", "opus", 48000);
 	set_next_mime_type(ast_format_vp8, 0,  "video", "VP8", 90000);
 	set_next_mime_type(ast_format_vp9, 0, "video", "VP9", 90000);
+	set_next_mime_type(ast_format_vevs, 0, "audio", "EVS", 16000);
 
 	/* Define the static rtp payload mappings */
 	add_static_payload(0, ast_format_ulaw, 0);
diff --git a/makeopts.in b/makeopts.in
index f7824f3476..f3eef0494e 100644
--- a/makeopts.in
+++ b/makeopts.in
@@ -351,4 +351,7 @@ SNDFILE_LIB=@SNDFILE_LIB@
 BEANSTALK_INCLUDE=@BEANSTALK_INCLUDE@
 BEANSTALK_LIB=@BEANSTALK_LIB@
 
+VEVS_INCLUDE=@VEVS_INCLUDE@
+VEVS_LIB=@VEVS_LIB@
+
 HAVE_SBIN_LAUNCHD=@PBX_LAUNCHD@
diff --git a/res/res_format_attr_vevs.c b/res/res_format_attr_vevs.c
new file mode 100644
index 0000000000..89b9d8fe28
--- /dev/null
+++ b/res/res_format_attr_vevs.c
@@ -0,0 +1,152 @@
+/*** MODULEINFO
+	<support_level>core</support_level>
+ ***/
+
+#include "asterisk.h"
+
+#include <ctype.h>                      /* for tolower */
+
+#include "asterisk/module.h"
+#include "asterisk/format.h"
+#include "asterisk/logger.h"            /* for ast_log, LOG_WARNING */
+#include "asterisk/strings.h"           /* for ast_str_append */
+#include "asterisk/utils.h"             /* for MAX, MIN */
+
+struct evs_attr {
+	int dummy;
+};
+
+static void evs_destroy(struct ast_format *format)
+{
+	struct evs_attr *attr = ast_format_get_attribute_data(format);
+
+	ast_free(attr);
+}
+
+static void attr_init(struct evs_attr *attr)
+{
+	memset(attr, 0, sizeof(*attr));
+}
+
+static int evs_clone(const struct ast_format *src, struct ast_format *dst)
+{
+	struct evs_attr *original = ast_format_get_attribute_data(src);
+	struct evs_attr *attr = ast_malloc(sizeof(*attr));
+
+#warning hacking
+abort();
+	if (!attr) {
+		return -1;
+	}
+
+	if (original) {
+		*attr = *original;
+	} else {
+		attr_init(attr);
+	}
+
+	ast_format_set_attribute_data(dst, attr);
+
+	return 0;
+}
+
+static struct ast_format *evs_parse_sdp_fmtp(const struct ast_format *format, const char *attributes)
+{
+	char *attribs = ast_strdupa(attributes), *attrib;
+	struct ast_format *cloned;
+	struct evs_attr *attr;
+//	unsigned int val;
+
+	cloned = ast_format_clone(format);
+	if (!cloned) {
+		return NULL;
+	}
+	attr = ast_format_get_attribute_data(cloned);
+
+	/* lower-case everything, so we are case-insensitive */
+	for (attrib = attribs; *attrib; ++attrib) {
+		*attrib = tolower(*attrib);
+	} /* based on channels/chan_sip.c:process_a_sdp_image() */
+
+	ast_log(LOG_WARNING, "Unhandled received attribute '%s', please fix!\n", attrib);
+
+	return cloned;
+}
+
+static void evs_generate_sdp_fmtp(const struct ast_format *format, unsigned int payload, struct ast_str **str)
+{
+	struct evs_attr *attr = ast_format_get_attribute_data(format);
+
+#warning hacking
+//	if (!attr) {
+//		return;
+//	}
+
+	ast_str_append(str, 0, "a=fmtp:%u br=5.9; bw=nb-wb\r\n", payload);
+}
+
+static enum ast_format_cmp_res evs_cmp(const struct ast_format *format1, const struct ast_format *format2)
+{
+	if (ast_format_get_sample_rate(format1) == ast_format_get_sample_rate(format2)) {
+		return AST_FORMAT_CMP_EQUAL;
+	}
+
+	return AST_FORMAT_CMP_NOT_EQUAL;
+}
+
+static struct ast_format *evs_getjoint(const struct ast_format *format1, const struct ast_format *format2)
+{
+	struct evs_attr *attr1 = ast_format_get_attribute_data(format1);
+	struct evs_attr *attr2 = ast_format_get_attribute_data(format2);
+	struct ast_format *jointformat;
+	struct evs_attr *attr_res;
+
+	if (ast_format_get_sample_rate(format1) != ast_format_get_sample_rate(format2)) {
+		return NULL;
+	}
+
+	jointformat = ast_format_clone(format1);
+	if (!jointformat) {
+		return NULL;
+	}
+	attr_res = ast_format_get_attribute_data(jointformat);
+
+	if (!attr1 || !attr2) {
+		attr_init(attr_res);
+	} else {
+		/* Take the lowest max bitrate */
+//		attr_res->maxbitrate = MIN(attr1->maxbitrate, attr2->maxbitrate);
+	}
+
+	return jointformat;
+}
+
+static struct ast_format_interface evs_interface = {
+	.format_destroy = evs_destroy,
+	.format_clone = evs_clone,
+	.format_cmp = evs_cmp,
+	.format_get_joint = evs_getjoint,
+	.format_parse_sdp_fmtp = evs_parse_sdp_fmtp,
+	.format_generate_sdp_fmtp = evs_generate_sdp_fmtp,
+};
+
+static int load_module(void)
+{
+	if (ast_format_interface_register("vevs", &evs_interface)) {
+		return AST_MODULE_LOAD_DECLINE;
+	}
+
+	return AST_MODULE_LOAD_SUCCESS;
+}
+
+static int unload_module(void)
+{
+	return 0;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Vocal EVS Format Attribute Module",
+	.support_level = AST_MODULE_SUPPORT_CORE,
+	.load = load_module,
+	.unload = unload_module,
+	.load_pri = AST_MODPRI_CHANNEL_DEPEND,
+);