Log Message:
Update to 0.0.17:

    Version 0.0.17

    tests: Add test for telephone-event events parameter nego

    rtpspecificnego: Add handling of telephone-event event ranges

    tests: Skip tests if no local candidates are produced

    rtcpfilter: Reduce the packet size when reducing the packet

    tests: Skip libnice tests if it finds no local candidates

    rtpdtmfsoundsource: Respect the ptime/maxptime too

    tests: Add test ptime/maxptime passing

    rtpsession: Set the ptime/maxptime on the send codec bin caps

    rtpcodecnego: Negotiate the ptime/maxptime

    rtpconference: Add function to make gst caps while keeping the ptime

    rtpcodecnego: Add function to copy the list of codecs with the send-side ptime

    tests; Add test for fscodec ptime/maxptime handling

    codec: Add ptime

    codec: Add maxptime

    tests: Take rtpsession lock during message emissions
    This ensures that it is not held across message emissions.

    tests: Add debug-blocks

    rtpsubstream: Keep ref on substream while callbacks are invoked

    rtpsubstream: Put codec/codecbin inside loop

    rtpsubstream: Use rw-lock to make sure the substream really stops

    rtp: Move locking into callback

    rtpsubstream: Don't hold session lock too much while setting new codecbin

    rtpsubstream: Move modification locking to blocked function
    Also allow only one thread to be in substream blocked function at once.

    rtp: Move substream blocking logic into substream

    rtp: Don't include marshaller headers in headers

    rtp: Depend on the correct var for marshaller list generation

    rtcpfilter: Add gst-p-base paths to Makefile.am
    Patch contributed by Armijn Hemel <armijn@loohuis-consulting.nl>

    rawudp: Remove upnp-request-timeout, it was a terrible idea

    Substitute deprecated Glib symbol: g_mapped_file_free
    Use g_mapped_file_unref if Glib >= 2.22 is available
    http://bugs.freedesktop.org/show_bug.cgi?id=21422

    rtpsession: Only add stream to list if its creation worked

    README: Require gst-p-bad 0.10.17 for dtmfsrc
    dtmfsrc can do do more than 8000 Hz, that has only been fixed in
    gst-plugins-bad 0.10.17

    rtpdtmfsound: Try hardwired PCMx only if the clock-rate is 8000

    rtp: Lookup codec with config is always for sending, so make it explicit
    Also, the dtmf sound will always get a valid codec now.

    rtpconference: Make message about gst_bin_add failure more accurate

    rtpdtmfsoundsource: Ignore codecs that don't have a blueprint

    tests: Test dtmf as sound

    tests: Make recv-pipeline per test

    rtpdtmfsoundsource: Use main codec if PCMA/U are not available

    rtpspecialsource: Make local class_get_codec function static

    rtp: Regroup CodecBlueprint related functions in one place

    rtpspecialsource: Rename negotiated_codecs to negotiated_codec_associations
    This way, the list contents can be guessed

    rtpsession: Don't need to set queue-delay anymore

    rtpsession: Split codecbin generation from factory from profile

    tests: Make it build against GUPnP 0.13

    msnsession: Check if dispose has already been called

    fstransmitter: uint can't be < 0

    rawudp: Bring upnp discovery timeout down to 2 seconds

    tests: Verify that it is not possible to disable all codecs
    Add a reserve-pt to guarantee that it is not possible to disable all codecs

    rtpcodecnego: Verify if there are any valid local codecs left after applying \ 
preferences

    rtpsession: Make error message less cryptic

    Version 0.0.16.1