KTp/Tasks/NewCallUI: Difference between revisions

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Note: leonhandreke from IRC discovered there's a bug in farstream-0.2 0.2.2 , you'll need at least 0.2.3 for video to work.
Note: leonhandreke from IRC discovered there's a bug in farstream-0.2 0.2.2 , you'll need at least 0.2.3 for video to work.
===Runtime Dependencies===
The set of available codecs depends on what gstreamer 1.0+ components you have installed. gstreamer1.0-plugins-good should give you enough for audio.
However if you want to do video chats with Jitsi you'll need H.264 support, that is probably available in gstreamer-1.0 libav.


===Checking you've done it right===
===Checking you've done it right===

Revision as of 22:49, 5 June 2014

About

Upstream upgraded from Farsight 0.1 to Farsight 0.2

Farsight is a library that manages codec negotiation in telepathy calls.

Farsight 0.1 is built against GStreamer0.10 Farsight 0.2 is built against GStreamer1.0


Farsight 0.1 is old and broken.

In order for us to use Farsight 0.2 we need GStreamer1.0 which means we need QtGStreamer 1.0.

QtGStreamer 1.0 had the annoying problem of not existing. Diane has ported it \o/

Building

(Diane's work at https://github.com/detrout/qt-gstreamer has been merged into the freedesktop master, so it's probably better to use that version)

Install QtGStreamer from http://cgit.freedesktop.org/gstreamer/qt-gstreamer/

git clone http://cgit.freedesktop.org/gstreamer/qt-gstreamer/
mkdir build
cd build
cmake -DQTGSTREAMER_CODEGEN=ON -DQTGSTREAMER_TESTS=ON ..
make

Recompile TpQt from https://github.com/davidedmundson/telepathy-qt Branch farstream-0.2-port

Recompile ktp-call-ui (normal KDE repo) branch gst-1.0-port

Note: leonhandreke from IRC discovered there's a bug in farstream-0.2 0.2.2 , you'll need at least 0.2.3 for video to work.

Runtime Dependencies

The set of available codecs depends on what gstreamer 1.0+ components you have installed. gstreamer1.0-plugins-good should give you enough for audio.

However if you want to do video chats with Jitsi you'll need H.264 support, that is probably available in gstreamer-1.0 libav.

Checking you've done it right

Run

ldd /opt/kde4/lib/kde4/libexec/ktp-call-ui | grep -i gstre

and make sure everything is at 1.0.

If you see 0.10, you've done it wrong.

Troubleshooting

SIP and Jingle are complex protocols and the software stack involved in making this work is fairly complex, so things are likely to go wrong.

ktp-call-ui is interacting with telepathy and farstream. Farstream is a set of plugins for gstreamer that provides support for RTP channels and codec negotiation. Somehow it calls libnice to do the NAT traversal.

You can watch ktp-call-ui interact with telepathy and the SIP or XMPP connection manager using ktp-debugger.

I've found you can do

export GST_DEBUG=fsrtpconference_disco:5,fsrtpconference_nego:5
${KDELIB}kde4/libexec//ktp-call-ui --persist

To see some of the underlying negotiation being done by farstream. (5 is DEBUG level).

Work that needs doing

Status Action Notes Developer
DONE Fix QtGStreamer pkgconfig files Merged David's patches <detrout>
TODO Fix QtGStreamer examples <detrout>
DONE Merge QtGStreamer 1.0-proposed None <detrout>
TODO Release QtGStreamer None <detrout>
IN PROGRESS Patch TpQt (done in branch) <[email protected]>
TODO Release TpQt None <[email protected]>
TODO Merge KTp-Call-UI Branch None <[email protected]>