--- include/asterisk/format.h
+++ include/asterisk/format.h
@@ -219,6 +219,10 @@
 	 * \retval -1 failure, Value was either not found, or not allowed to be accessed.
 	 */
 	int (* const format_attr_get_val)(const struct ast_format_attr *format_attr, int key, void *val);
+	
+	int (* const format_samples)(const struct ast_frame *f);
+	int (* const format_rate)(const struct ast_format *format);
+	int (* const allowSmoother)(void);
 
 	/*
 	 * \brief Parse SDP attribute information, interpret it, and store it in ast_format_attr structure.
@@ -474,1 +475,7 @@
+
+int ast_format_dynamic_add(struct ast_format_list *new);
+int ast_format_dynamic_get_samples(const struct ast_format *format, const struct ast_frame *frame);
+int ast_format_dynamic_get_rate(const struct ast_format *format);
+int ast_format_dynamic_register(struct ast_format_attr_interface *new, enum ast_format_type formatType);
+int ast_format_allowSmoother(const struct ast_format *format);
 #endif /* _AST_FORMAT_H */
--- include/asterisk/rtp_engine.h
+++ include/asterisk/rtp_engine.h
@@ -2134,6 +2134,8 @@
  * \param dtls_cfg a DTLS configuration structure
  */
 void ast_rtp_dtls_cfg_free(struct ast_rtp_dtls_cfg *dtls_cfg);
+	
+void ast_rtp_format_dynamic_add(const struct ast_format *format, int rtp_code, char *type, char *subtype, unsigned int sample_rate, int map);
 
 #if defined(__cplusplus) || defined(c_plusplus)
 }
--- main/channel.c
+++ main/channel.c
@@ -928,6 +928,7 @@
 	};
 	char buf[512];
 	int x;
+	unsigned int found_audio = 0;
 
 	/* Find the first preferred codec in the format given */
 	for (x = 0; x < ARRAY_LEN(prefs); x++) {
@@ -935,6 +936,25 @@
 			return result;
 		}
 	}
+	
+	/* if just one format is audio, we know the best one */
+	ast_format_cap_iter_start(cap);
+	while (!ast_format_cap_iter_next(cap, result)) {
+		if (AST_FORMAT_GET_TYPE(result->id) == AST_FORMAT_TYPE_AUDIO) {
+			found_audio++;
+		}
+	}
+	ast_format_cap_iter_end(cap);
+	if (1 == found_audio) {
+		ast_format_cap_iter_start(cap);
+		while (!ast_format_cap_iter_next(cap, result)) {
+			if (AST_FORMAT_GET_TYPE(result->id) == AST_FORMAT_TYPE_AUDIO) {
+				break;
+			}
+		}
+		ast_format_cap_iter_end(cap);
+		return result;
+	}
 
 	ast_format_clear(result);
 	ast_log(LOG_WARNING, "Don't know any of %s formats\n", ast_getformatname_multiple(buf, sizeof(buf), cap));
--- main/format.c
+++ main/format.c
@@ -172,7 +172,7 @@
 	struct interface_ao2_wrapper *wrapper;
 
 	if (!(wrapper = find_interface(format))) {
-		ast_log(LOG_WARNING, "Could not find format interface to set.\n");
+		ast_log(LOG_WARNING, "Could not find format interface for %d to set.\n", format->id);
 		return -1;
 	}
 
@@ -729,6 +729,8 @@
 
 int ast_format_rate(const struct ast_format *format)
 {
+	int rate;
+	
 	switch (format->id) {
 	case AST_FORMAT_SLINEAR12:
 		return 12000;
@@ -783,7 +785,8 @@
 		}
 	}
 	default:
-		return 8000;
+		rate = ast_format_dynamic_get_rate(format);
+		return (rate > 0) ? rate : 8000;
 	}
 }
 
@@ -1427,3 +1426,102 @@
 	f_list = ast_format_list_destroy(f_list);
 	return 0;
 }
+
+int ast_format_dynamic_add(struct ast_format_list *new)
+{
+	RAII_VAR(struct ast_format_list *, entry, NULL, ao2_cleanup);
+	if (!(entry = ao2_alloc(sizeof(*entry), NULL))) {
+		return -1;
+	}
+	if (new->custom_entry != 0) {
+		/*
+		 * TODO load_format_config:ao2_callback is called at 
+		 * startup, several times. That removes all custom 
+		 * formats. If you look for this feature, please,
+		 * contact the author of this patch. Workaround:
+		 * via the command-line interface, core unload the
+		 * module, core load it again after each restart.
+		 */
+		ast_log(LOG_WARNING, "Custom format; is removed at restart.\n");
+	}
+	memcpy(entry, new, sizeof(struct ast_format_list));
+	/* keeps custom_entry as specified, default = 0 = static */
+	ao2_link(format_list, entry);
+	
+	return build_format_list_array();
+}
+
+int ast_format_dynamic_get_samples(const struct ast_format *format, const struct ast_frame *f)
+{
+	struct interface_ao2_wrapper *wrapper;
+	int samples = 0;
+	
+	if ((wrapper = find_interface(format))) {
+		if (wrapper->interface && wrapper->interface->format_samples) {
+			samples = wrapper->interface->format_samples(f);
+		}
+		ao2_ref(wrapper, -1);
+	}
+	
+	return samples;
+}
+
+int ast_format_dynamic_get_rate(const struct ast_format *format)
+{
+	struct interface_ao2_wrapper *wrapper;
+	int rate = -1;
+	
+	if ((wrapper = find_interface(format))) {
+		if (wrapper->interface && wrapper->interface->format_rate) {
+			rate = wrapper->interface->format_rate(format);
+		}
+		ao2_ref(wrapper, -1);
+	}
+	
+	return rate;
+}
+
+int ast_format_allowSmoother(const struct ast_format *format)
+{
+	struct interface_ao2_wrapper *wrapper;
+	int result = 1;
+	
+	if ((wrapper = find_interface(format))) {
+		if (wrapper->interface && wrapper->interface->allowSmoother) {
+			result = wrapper->interface->allowSmoother();
+		}
+		ao2_ref(wrapper, -1);
+	}
+	
+	return result;
+}
+
+int ast_format_dynamic_register(struct ast_format_attr_interface *newFormatInterface, enum ast_format_type formatType)
+{
+	size_t x, f_len;
+	const struct ast_format_list *f_list;
+	
+	if (newFormatInterface->id == 0) {
+		/*
+		 * CELT is the highest (audio) format as of Asterisk 11, and
+		 * CELT might not be in the format list because is not added statically,
+		 * or stated differently: CELT is the minimum ID (see format.h).
+		 * Not an issue since Asterisk 12 because of OPUS which is even higher.
+		 */
+		newFormatInterface->id = AST_FORMAT_CELT + 1;
+		
+		ao2_wrlock(interfaces);
+		f_list = ast_format_list_get(&f_len);
+		for (x = 0; x < f_len; x++) {
+			if (AST_FORMAT_GET_TYPE(f_list[x].format.id) == formatType) {
+				if (newFormatInterface->id <= f_list[x].format.id) {
+					newFormatInterface->id = f_list[x].format.id + 1;
+				}
+			}
+		}
+		f_list = ast_format_list_destroy(f_list);
+		ao2_unlock(interfaces);
+	}
+	
+	return ast_format_attr_reg_interface(newFormatInterface);
+}
--- main/frame.c
+++ main/frame.c
@@ -1084,7 +1084,10 @@
 		samples = ast_format_rate(&f->subclass.format) / 50;
 		break;
 	default:
-		ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(&f->subclass.format));
+		samples = ast_format_dynamic_get_samples(&f->subclass.format, f);
+		if (!samples) {
+			ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(&f->subclass.format));
+		}
 	}
 	return samples;
 }
--- main/rtp_engine.c
+++ main/rtp_engine.c
@@ -2370,3 +2370,9 @@
 
 	return 0;
 }
+
+void ast_rtp_format_dynamic_add(const struct ast_format *format, int rtp_code, char *type, char *subtype, unsigned int sample_rate, int map)
+{
+	set_next_mime_type(format, rtp_code, type, subtype, sample_rate);
+	add_static_payload(map, format, rtp_code);
+}
--- res/res_rtp_asterisk.c
+++ res/res_rtp_asterisk.c
@@ -3000,6 +3000,7 @@
 	struct ast_sockaddr remote_address = { {0,} };
 	struct ast_format subclass;
 	int codec;
+	int allowSmoother = 1;
 
 	ast_rtp_instance_get_remote_address(instance, &remote_address);
 
@@ -3046,9 +3046,7 @@
 		}
 	}
 
-	/* If no smoother is present see if we have to set one up */
-	if (!rtp->smoother) {
-		struct ast_format_list fmt = ast_codec_pref_getsize(&ast_rtp_instance_get_codecs(instance)->pref, &subclass);
+		allowSmoother = ast_format_allowSmoother(&subclass);
 
 		switch (subclass.id) {
 		case AST_FORMAT_SPEEX:
@@ -3062,8 +3060,15 @@
 		case AST_FORMAT_G719:
 			/* these are all frame-based codecs and cannot be safely run through
 			   a smoother */
+			allowSmoother = 0;
 			break;
 		default:
+			break;
+		}
+		
+		/* If no smoother is present see if we have to set one up */
+		if (!rtp->smoother && allowSmoother) {
+			struct ast_format_list fmt = ast_codec_pref_getsize(&ast_rtp_instance_get_codecs(instance)->pref, &subclass);
 			if (fmt.inc_ms) {
 				if (!(rtp->smoother = ast_smoother_new((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms))) {
 					ast_log(LOG_WARNING, "Unable to create smoother: format %s ms: %d len: %d\n", ast_getformatname(&subclass), fmt.cur_ms, ((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms));
@@ -3075,7 +3080,6 @@
 				ast_debug(1, "Created smoother: format: %s ms: %d len: %d\n", ast_getformatname(&subclass), fmt.cur_ms, ((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms));
 			}
 		}
-	}
 
 	/* Feed audio frames into the actual function that will create a frame and send it */
 	if (rtp->smoother) {