Merge "pjsip: Extend 'asymmetric_rtp_codec' option to include us changing." into 13

CHANGES

@@ -34,6 +34,12 @@ chan_pjsip
    function any contact which is considered unreachable due to qualify being
    enabled will no longer be called.
 
+ * The asymmetric_rtp_codec option now also controls whether chan_pjsip will
+   send media as-is without transcoding if the codec has been negotiated in the
+   SDP. If set to "no" then Asterisk will only ever send the preferred codec
+   from the SDP, unless the remote side sends a different codec and we will
+   switch to match.
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 13.15.0 to Asterisk 13.16.0 ----------
 ------------------------------------------------------------------------------

channels/chan_pjsip.c

@@ -737,11 +737,24 @@ static struct ast_frame *chan_pjsip_read(struct ast_channel *ast)
 
 	if (!session->endpoint->asymmetric_rtp_codec &&
 		ast_format_cmp(ast_channel_rawwriteformat(ast), f->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) {
-		/* For maximum compatibility we ensure that the write format matches that of the received media */
+		struct ast_format_cap *caps;
+
+		/* For maximum compatibility we ensure that the formats match that of the received media */
 		ast_debug(1, "Oooh, got a frame with format of %s on channel '%s' when we're sending '%s', switching to match\n",
 			ast_format_get_name(f->subclass.format), ast_channel_name(ast),
 			ast_format_get_name(ast_channel_rawwriteformat(ast)));
+
+		caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
+		if (caps) {
+			ast_format_cap_append_from_cap(caps, ast_channel_nativeformats(ast), AST_MEDIA_TYPE_UNKNOWN);
+			ast_format_cap_remove_by_type(caps, AST_MEDIA_TYPE_AUDIO);
+			ast_format_cap_append(caps, f->subclass.format, 0);
+			ast_channel_nativeformats_set(ast, caps);
+			ao2_ref(caps, -1);
+		}
+
 		ast_set_write_format_path(ast, ast_channel_writeformat(ast), f->subclass.format);
+		ast_set_read_format_path(ast, ast_channel_readformat(ast), f->subclass.format);
 
 		if (ast_channel_is_bridged(ast)) {
 			ast_channel_set_unbridged_nolock(ast, 1);

res/res_pjsip_sdp_rtp.c

@@ -401,7 +401,24 @@ static int set_caps(struct ast_sip_session *session, struct ast_sip_session_medi
 		ast_format_cap_append_from_cap(caps, ast_channel_nativeformats(session->channel),
 			AST_MEDIA_TYPE_UNKNOWN);
 		ast_format_cap_remove_by_type(caps, media_type);
-		ast_format_cap_append_from_cap(caps, joint, media_type);
+
+		/*
+		 * If we don't allow the sending codec to be changed on our side
+		 * then get the best codec from the joint capabilities of the media
+		 * type and use only that. This ensures the core won't start sending
+		 * out a format that we aren't currently sending.
+		 */
+		if (!session->endpoint->asymmetric_rtp_codec) {
+			struct ast_format *best;
+
+			best = ast_format_cap_get_best_by_type(joint, media_type);
+			if (best) {
+				ast_format_cap_append(caps, best, ast_format_cap_get_framing(joint));
+				ao2_ref(best, -1);
+			}
+		} else {
+			ast_format_cap_append_from_cap(caps, joint, media_type);
+		}
 
 		/*
 		 * Apply the new formats to the channel, potentially changing