[gst-devel] gstreamer develop---stream a mp3 file via rtsp & rtp

[聂巍]

I use gstreamer to stream a mp3 file via rtsp & rtp
and I only heard the sound for about half a second and it stops at once

The console displays :
WARNING: Element "alsasink0" warns:
gstbaseaudiosink.c(696): gst_base_audio_sink_render ():
/pipeline0/alsasink0:
Unexpected discontinuity in audio timestamps of more than half a second
(0:00:00.803061224), resyncing


My commadline is : gst-launch rtspsrc location =
"rtsp://192.168.1.10/testStream.mp3" ! rtpmpadepay ! queue ! mad ! alsasink

would anybody tell me where the problem is ?
and how can i solve it ~~
thank you guys in advance :)
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