KTp/Tasks/NewCallUI: Difference between revisions
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Farsight 0.1 is built against GStreamer0.10 | Farsight 0.1 is built against GStreamer0.10 | ||
Farsight 0.2 is built against GStreamer1.0 | Farsight 0.2 is built against GStreamer1.0 | ||
Farsight 0.1 is old and broken. | Farsight 0.1 is old and broken. | ||
| Line 18: | Line 17: | ||
===Building=== | ===Building=== | ||
You will need at least Qt Gstreamer 1.2. If it's not available for your distribution you can obtain it from http://cgit.freedesktop.org/gstreamer/qt-gstreamer/ If you need to build it you probably should use the example cmake command line below. | |||
You need at least farstream 0.2.3, telepathy-farstream 0.6 and telepathy-qt 0.9.5 | |||
{{Input|1=<nowiki> | {{Input|1=<nowiki> | ||
git clone http://cgit.freedesktop.org/gstreamer/qt-gstreamer | git clone http://cgit.freedesktop.org/gstreamer/qt-gstreamer | ||
cd qt-gstreamer | |||
mkdir build | mkdir build | ||
cd build | cd build | ||
| Line 29: | Line 29: | ||
make</nowiki>}} | make</nowiki>}} | ||
As of 2015 Mar 18, ktp-call-ui (normal KDE repo) master contains the KDE4 version. A Plasma 5 version one should be done soon. | |||
===Checking you've done it right=== | ===Checking you've done it right=== | ||
| Line 44: | Line 40: | ||
and make sure everything is at 1.0. | and make sure everything is at 1.0. | ||
If you see 0.10, you've done | If you see a 0.10 version of gstreamer anywhere, something went wrong. | ||
===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. | |||
===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 | |||
{{Input|1=<nowiki> | |||
export GST_DEBUG=fsrtpconference_disco:5,fsrtpconference_nego:5 | |||
${KDELIB}kde4/libexec//ktp-call-ui --persist | |||
</nowiki>}} | |||
To see some of the underlying negotiation being done by farstream. (5 is DEBUG level). | |||
There can also be issues trying to negotiate with SIP servers, to debug your SIP connection, turn off your sip accounts in the KDE telepathy account list, and then do the following in a terminal | |||
{{Input|1=<nowiki> | |||
export TPORT_LOG=1 | |||
export TPORT_DUMP=/tmp/sip.log | |||
export RAKIA_DEBUG=all | |||
/usr/lib/telepathy/telepathy-rakia | |||
</nowiki>}} | |||
Once you reactivate the SIP account, the SIP protocol messages should be then written to /tmp/sip.log. | |||
More information about setting up rakia/sophia debugging can be found at http://cgit.freedesktop.org/telepathy/telepathy-rakia/tree/README | |||
===Work that needs doing=== | ===Work that needs doing=== | ||
| Line 59: | Line 91: | ||
{{FeatureProjectTodo|Fix QtGStreamer examples||detrout}} | {{FeatureProjectTodo|Fix QtGStreamer examples||detrout}} | ||
{{FeatureProjectDone|Merge QtGStreamer 1.0-proposed|None|detrout}} | {{FeatureProjectDone|Merge QtGStreamer 1.0-proposed|None|detrout}} | ||
{{ | {{FeatureProjectDone|Release QtGStreamer|None|detrout}} | ||
{{ | {{FeatureProjectDone|Patch TpQt|(done in branch)|[email protected]}} | ||
{{ | {{FeatureProjectDone|Release TpQt|None|[email protected]}} | ||
{{ | {{FeatureProjectDone|Merge KTp-Call-UI Branch|None|[email protected]}} | ||
{{FeatureProjectDone|Merge Kaditx's [https://gitorious.org/kde-telepathy-kaditx/ktp-call-newui QML Call UI]||}} | |||
{{FeatureProjectTodo|Port to Frameworks 5|None|olesalscheider}} | |||
{{FeatureProjectTodo|Continue debugging SIP call issues|None|detrout}} | |||
|} | |} | ||
Latest revision as of 23:07, 1 June 2015
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
You will need at least Qt Gstreamer 1.2. If it's not available for your distribution you can obtain it from http://cgit.freedesktop.org/gstreamer/qt-gstreamer/ If you need to build it you probably should use the example cmake command line below.
You need at least farstream 0.2.3, telepathy-farstream 0.6 and telepathy-qt 0.9.5
git clone http://cgit.freedesktop.org/gstreamer/qt-gstreamer cd qt-gstreamer mkdir build cd build cmake -DQTGSTREAMER_CODEGEN=ON -DQTGSTREAMER_TESTS=ON .. make
As of 2015 Mar 18, ktp-call-ui (normal KDE repo) master contains the KDE4 version. A Plasma 5 version one should be done soon.
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 a 0.10 version of gstreamer anywhere, something went wrong.
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.
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).
There can also be issues trying to negotiate with SIP servers, to debug your SIP connection, turn off your sip accounts in the KDE telepathy account list, and then do the following in a terminal
export TPORT_LOG=1 export TPORT_DUMP=/tmp/sip.log export RAKIA_DEBUG=all /usr/lib/telepathy/telepathy-rakia
Once you reactivate the SIP account, the SIP protocol messages should be then written to /tmp/sip.log.
More information about setting up rakia/sophia debugging can be found at http://cgit.freedesktop.org/telepathy/telepathy-rakia/tree/README
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> |
| DONE | Release QtGStreamer | None | <detrout> |
| DONE | Patch TpQt | (done in branch) | <[email protected]> |
| DONE | Release TpQt | None | <[email protected]> |
| DONE | Merge KTp-Call-UI Branch | None | <[email protected]> |
| DONE | Merge Kaditx's QML Call UI | ||
| TODO | Port to Frameworks 5 | None | <olesalscheider> |
| TODO | Continue debugging SIP call issues | None | <detrout> |