--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -392,7 +392,10 @@ static int set_caps(struct ast_sip_sessi
 	if (!ast_format_cap_count(joint)) {
 		struct ast_str *usbuf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
 		struct ast_str *thembuf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
-
+		if(codecs.framing) {
+			/* Copy the codecs framing ms even though there is no match */
+			ast_rtp_codecs_set_framing(ast_rtp_instance_get_codecs(session_media->rtp),codecs.framing);
+		}
 		ast_rtp_codecs_payloads_destroy(&codecs);
 		ast_log(LOG_NOTICE, "No joint capabilities for '%s' media stream between our configuration(%s) and incoming SDP(%s)\n",
 			session_media->stream_type,
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -5157,6 +5157,13 @@ static int bridge_p2p_rtp_write(struct a
 		return -1;
 	}
 
+	/* In p2p mode the framing ms values for both must be equal. If bridged is zero we don't care */
+	unsigned framing1 = ast_rtp_codecs_get_framing(ast_rtp_instance_get_codecs(instance1));
+	unsigned framing = ast_rtp_codecs_get_framing(ast_rtp_instance_get_codecs(instance));
+	if (framing1 != 0 && framing != framing1) {
+		return -1;
+	}
+
 	/* Otherwise adjust bridged payload to match */
 	bridged_payload = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance1), payload_type->asterisk_format, payload_type->format, payload_type->rtp_code);