diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index 43002caf38..77b59a3ff3 100644
--- a/libavformat/rtpdec.c
+++ b/libavformat/rtpdec.c
@@ -393,7 +393,6 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *r
             return NULL;
         }
     } else {
-        av_set_pts_info(st, 32, 1, 90000);
         switch(st->codec->codec_id) {
         case CODEC_ID_MPEG1VIDEO:
         case CODEC_ID_MPEG2VIDEO:
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index 1dd166e3f1..2703400720 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -333,7 +333,8 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
                 RTPDynamicProtocolHandler *handler;
                 /* if standard payload type, we can find the codec right now */
                 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
-                if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO)
+                if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
+                    st->codec->sample_rate > 0)
                     av_set_pts_info(st, 32, 1, st->codec->sample_rate);
                 /* Even static payload types may need a custom depacketizer */
                 handler = ff_rtp_handler_find_by_id(