transport: Add parsers before decoders and after encoders

This is in general required in 1.x and in specific if different stream formats
or parsed information are needed. For us the only format that has these
restrictions is H264 (and H265/HEVC later, or AAC if ever added).
transport: Don't shadow local variables
transport: Move valve from before the decoder to afterwards

If we have the valve before the decoder we will drop all
encoded frames until the stream is linked, including the
initial keyframe usually. So we have to request a new
keyframe which might or might not work but in any case
will take some extra time and delay the initial setup
of the pipeline.
owr: Constify some static arrays
transport: Use the highest ranked, working decoder/encoder for VP8/H264 from the registry

This allows us to use existing codecs on the user's system, especially it
allows to use hardware codecs and removes a dependency on openh264.
configure: Properly align --enable-debug help string
owr: Conditionally build a static binary or not

Make the default no, so that a normal configure will work as expected
and builds a version that works out of the box with the system dependencies.

https://github.com/EricssonResearch/openwebrtc/issues/6
webrtc.js: handle invalid ICE candidates gracefully

On invalid ICE candidates invoke the failureCallback passed to addIceCandidate.
Also call completeQueuedOperation() so that the internal operation queue can
proceed and doesn't get stuck.
webrtc.js: Work around Chrome WebRTC bug #1142

https://code.google.com/p/webrtc/issues/detail?id=1142

Fixes interop with Chrome Android v37.0.2062.117, for example.

The RTCIceCandidate.candidate attribute as passed by Chrome for Android is of
the format `a=candidate-attribute', meaning that the actual candidate is
prefixed by a=.
Update scripts to include libnice 0.1.8
Update scripts to include libnice synchronous name resolution

This should be removed when it is better-implemented elsewhere.
transport: Always set up RTCP mux and normal RTCP pipelines and switch dynamically

Previously we were only setting up the non-RTCP mux pipeline unless
rtcp-mux was TRUE from the very beginning. This will never be the case
with WebRTC though, as we can only know if the other side can do RTCP mux
after SDPs are exchanged.

Now we always set up both paths, and for the sender dynamically switch
between RTCP muxing and normal RTCP depending on the property of the
media session. For receiving we just receive from both.
OwrTransportAgent: Add leaky queue before encoder

We don't want to queue up any large amount of buffers if downstream of
the encoder is blocked or if the encoder is taking too long to encode
frames. Also, we want the encoder to encode the most recent frame
available at the point when it is ready to encode the next frame.
Update scripts
webrtc_js: Fix incorrect casing in trickle ICE TCP candidate
Make also the related port be 9 for non-host TCP active candidates

In accordance with RFC 6544
peerhandler: Use the correct property name to enable/disable FIR
Don't create the RTCP component and elements when multiplexing RTP/RTCP

Also clarify the description of the rtcp-mux property

Closes #41
OwrImageRenderer: Add a queue to decouple conversion from decoding

Previously there was one thread from the jitterbuffer through to the
sink. Use max-size-buffers = 3 as the limiting factor to allow a bit of
scheduling room.
owr_utils: Remove chatty and not-useful pipeline debug log

This is too hard to read and dot graphs are much easier to use.