Need help with using OPUS over RTP

[Thomas DEBESSE]

For your information, I broadcast sound over the network with
GStreamer and Opus over RTP from alsa interface (as you try to do)
since Opus is available (I used CELT before). I mainly use alsasrc,
alsasink, jackaudiosrc, jackaudiosink, opussrc and opussink modules. I
broadcast continuously, and I meet problems every day due to an
alsasrc and alsasink bug, but the cause has not been identified (bug
appears on both Gstreamer 0.10 and Gstreamer 1.0).

If you intend to broadcast sound for several hours or even several
days using alsasrc or alsasink module, expect to encounter bugs
(silence, noise).

You can read the bug report :
https://bugzilla.gnome.org/show_bug.cgi?id=692953
You can also read the discussion on gstreamer-devel mail list :
http://gstreamer-devel.966125.n4.nabble.com/long-run-gstreamer-stops-working-without-message-td4655520.html

-- 
Thomas DEBESSE
RCF Méditerrannée