--- main/translate.c	(Asterisk 13.38.1)
+++ main/translate.c	(working copy)
@@ -1501,14 +1501,17 @@ static void check_translation_path(
 	enum ast_media_type type)
 {
-	int index, src_index = format2index(src_fmt);
+	int i;
+
+	if (ast_format_get_type(src_fmt) != type) {
+		return;
+	}
+
 	/* For a given source format, traverse the list of
 	   known formats to determine whether there exists
 	   a translation path from the source format to the
 	   destination format. */
-	for (index = 0; (src_index >= 0) && index < cur_max_index; index++) {
-		struct ast_codec *codec = index2codec(index);
-		RAII_VAR(struct ast_format *, fmt, ast_format_create(codec), ao2_cleanup);
-
-		ao2_ref(codec, -1);
+	for (i = ast_format_cap_count(result) - 1; 0 <= i; i--) {
+		int index, src_index;
+		struct ast_format *fmt = ast_format_cap_get_format(result, i);
 
 		if (ast_format_get_type(fmt) != type) {
@@ -1526,4 +1529,13 @@ static void check_translation_path(
 			continue;
 		}
+
+		/* if this is a pass-through format, not in the source,
+		   we cannot transcode. Therefore, remove it from the result */
+		src_index = format2index(src_fmt);
+		index = format2index(fmt);
+		if (src_index < 0 || index < 0) {
+			ast_format_cap_remove(result, fmt);
+			continue;
+		}
 
 		/* if we don't have a translation path from the src
--- channels/chan_sip.c	(Asterisk 13.38.1)
+++ channels/chan_sip.c	(working copy)
@@ -13546,7 +13546,4 @@ static enum sip_result add_sdp(struct si
 
-		/* Check if we need audio */
-		if (ast_format_cap_has_type(tmpcap, AST_MEDIA_TYPE_AUDIO)
-			|| ast_format_cap_has_type(p->caps, AST_MEDIA_TYPE_AUDIO)) {
-			needaudio = TRUE;
-		}
+		/* Check if we need audio in this call */
+		needaudio = ast_format_cap_has_type(tmpcap, AST_MEDIA_TYPE_AUDIO);
 
@@ -13681,3 +13678,2 @@ static enum sip_result add_sdp(struct si
 		   - Then our mutually shared capabilities, determined previous in tmpcap
-		   - Then preferences in order from sip.conf device config for this peer/user
 		*/
@@ -13725,23 +13721,2 @@ static enum sip_result add_sdp(struct si
 
-		/* Finally our remaining audio/video codecs */
-		for (x = 0; p->outgoing_call && x < ast_format_cap_count(p->caps); x++) {
-			tmp_fmt = ast_format_cap_get_format(p->caps, x);
-
-			if (ast_format_cap_iscompatible_format(alreadysent, tmp_fmt) != AST_FORMAT_CMP_NOT_EQUAL) {
-				ao2_ref(tmp_fmt, -1);
-				continue;
-			}
-
-			if (ast_format_get_type(tmp_fmt) == AST_MEDIA_TYPE_AUDIO) {
-				add_codec_to_sdp(p, tmp_fmt, &m_audio, &a_audio, debug, &min_audio_packet_size, &max_audio_packet_size);
-			} else if (needvideo && ast_format_get_type(tmp_fmt) == AST_MEDIA_TYPE_VIDEO) {
-				add_vcodec_to_sdp(p, tmp_fmt, &m_video, &a_video, debug, &min_video_packet_size);
-			} else if (needtext && ast_format_get_type(tmp_fmt) == AST_MEDIA_TYPE_TEXT) {
-				add_tcodec_to_sdp(p, tmp_fmt, &m_text, &a_text, debug, &min_text_packet_size);
-			}
-
-			ast_format_cap_append(alreadysent, tmp_fmt, 0);
-			ao2_ref(tmp_fmt, -1);
-		}
-
 		/* Now add DTMF RFC2833 telephony-event as a codec */