Problems with interleave element

[daniel at poradnik-webmastera.com]

Hi,
I have pipeline which sends audio from .wav file via RTP to VoIP server 
(with help of rtpbin element), and at the same time receives audio from 
it and saves it to another wav file. I wanted to measure delay 
introduced by VoIP server, by saving both input audio and output audio 
to separate channels of .wav file. To do this, I split decoded stream 
from .wav file into two using tee, and then send it to interleave 
element (plus necessary audioconvert/audiopresample/capsfilter) as 
channel 1. Channel 2 gets decoded rtp stream.

When I opened recorded file, I found that delay is about 5 secs, plus 
similar part of initial audio on channel 1 was truncated. GStreamer 
itself introduces small delay - on all 3 paths (filesrc->udpsink, 
udpsrc->filesink, filesrc->filesink) it is below 1ms. I also decoded 
captured RTP streams in Wireshark, and delay between them was 23ms. So 
things clearly do not sum up.

Looks that this is synchronization problem. I added sync=TRUE to 
filesink, what cut down about 1 sec. I also added audiomixer + 
audiotestsrc as silence generator after rtpbin and decoders to always 
have this part of pipeline connected - before I had to wait for signal 
from rtpbin that it added corresponding output pad. Now delay in 
recorded file is about 2 secs, what still is way too big.

Looks that my streams are still not properly synchronized. What am I 
missing here?

I also still see that initial audio truncation. I stop pipeline by 
changing state to NULL, so file is not closes properly and some trailing 
audio is lost. Is is possible that initial part can be lost too because 
of this?

BTW, I have just noticed that GStreamer provides interleave and 
audiointerleave elements, and it seems that both of them do the same 
thing. Why 2 similar elements exists?

Regards,
Daniel