[gst-devel] gst-launch rtp problem with filesink and packet loss

[Luca Gaggero]

Hi all,

I have a problem with a stream audio rtp.
I send from a pc1 to a pc2 a stream rtp with gst-launch, I use codec alaw.

In the sender I use:

gst-launch-0.10 -v filesrc location=./file.wav ! wavparse ! audioconvert 
! alawenc ! rtppcmapay ! udpsink host=192.168.1.2 port=5000

In the receiver I use:

gst-launch-0.10 -v udpsrc port=5000 caps="application/x-rtp, 
media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA,
     payload=(int)8" ! rtppcmadepay ! alawdec ! audioconvert ! 
audioresample ! wavenc  ! filesink location=output.wav sync=false

the stream arrive and if I listening the file output.wav it is correct.



But I need to introduce in the comunication a packet loss.
Now if I repeate the test with a packet loss the file is created, but 
the duration of the audio stream is less.
I want to write a silence frame when the packet are missing, as like as 
a listener listen in real time the audio stream...

Someone have a solution?
I also write a programm with java media framework but I have the same 
problem...