[Bug 720696] gst-rtsp-server: allow client to accept different transports

[GStreamer (bugzilla.gnome.org)]

https://bugzilla.gnome.org/show_bug.cgi?id=720696
  GStreamer | gst-rtsp-server | git

--- Comment #3 from Aleix Conchillo Flaqué <aleix at oblong.com> 2014-01-03 17:06:09 UTC ---
(In reply to comment #2)
> Or maybe there should be a deeper integration of RTP/SAVP inside
> gst-rtsp-server that takes care of inserting the srtp elements too?

Right now, I do that (inserting srtp elements) by subclassing rtsp-media and
implementing setup_rtpbin. There, I just connect to the new rtp-request-*
signals. The current gst-rtsp-server API is pretty flexible.

But it's true that, probably, everyone that wants srtp support will have to do
something similar, so it might be good to already add it to gst-rtsp-server.

In any case this allow transport callback is also flexible enough to let the
user choose anything whether gst-rtsp-server integrates srtp or not. I think it
might be useful for testing purposes or if you want to allow a certain
transport only in specific cases (e.g. config file, only certain URLs, IPs).

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