Ust GST_DEBUG_BIN_TO_DOT_FILE to generate a dot graph of a pipeline for
debugging purposes when SIGUSR2 is received.
Note the same signal is also used within the dummy plugin to simulate an
incoming call from an unknown number, so when testing you probably want either
the sip plugin or the dummy plugin, but not both.
We were using two distinct pipelines, one for receiving and one for
sending. The receive pipeline was set to the playing state to allocate
the sockets which we would reuse for the sending direction for our NAT
traversal scheme.
The rework to a single pipeline broke reusing sockets subtly.
This happened because the state of the GstUDPSrc could be reset leading
to newly allocated sockets once the pipeline is set to play.
This is now fixed by locking the state of the GstUDPSrc in the ready
state during socket reuse setup and while the pipeline is paused.
Additionally get rid of the "close-socket" property on the udp sources
because it was never needed.
Fixes aa446f82
sq
Using a single pipeline makes implementing encryption easier because we don't
need to duplicate srtpenc and srtpdec elements for each direction.
It also makes it easier to switch to using farstream down the line (see #426).
If the environment variable GST_DEBUG_DUMP_DOT_DIR is set, a graph of the send
and receive pipelines will be written to disk.
To generate a png from the exported dot files graphviz can be used like this:
`dot -Tpng -oimage.png graph.dot`
Now that initialization is split per pipeline and that the OS handles port
allocation we can move setting up socket reuse into the pipeline initialization
step instead of setting it up when starting the media pipelines.
This makes the calls_sip_media_pipeline_start() method a bit simpler.
We're also now reusing sockets for RTCP.
Closes#315
We're not setting the desired ports from the outside anymore, but rather
querying the ports that have been allocated by the operating system.
Therefore the lport-rtp and lport-rtcp property have become superfluous and are
being removed. We also adapt to changes outside of the pipeline code.
We don't expect the initialization to be able to fail. The only thing that could
potentially fail is setting up codecs and this has been delayed until after
initialization.
First of we get rid of the bindings between from "lport-rtp" and "lport-rtcp" to
the "port" property of the udpsrc elements. The properties themselves will get
removed a little later as the required changes are rather intrusive and we need
some more infrastructure in place before we can do the switch.
This is the first step in getting rid of the requirement to have the codec set
during object construction. The goal is to have pipelines prepared in advance so
that the codec can be plugged in once negotiation is complete.
Having the pipelines prepared in advance let's us grab allocated local ports of
udpsrc elements for RTP and RTCP instead of setting those and hoping they're not
yet in use.
They are phased out in favour of their newly introduced ui-call-* pendants.
This was done to have a better separation of concerns and allows for some
cleanup in CallsCall.
Closes#397
This function is used in the activate callback for the per protocol dial actions
to choose the correct origin to place a call from. If an origin cannot be found
it will return NULL which will lead to the fallback "app.dial" action being
invoked.
The id property will be used to keep track of which origin was used for a call,
so that we can default to reusing the same origin when placing a call from the
history.
Fixes the deprecation warning from meson:
DEPRECATION: target sip links against shared module sip, which is incorrect.
This will be an error in the future, so please use shared_library() for sip instead.
If shared_module() was used for sip because it has references to undefined symbols,
use shared_libary() with `override_options: ['b_lundef=false']` instead.
This gives us a better separation of concerns and it will make it a bit easier
to move the sip independent media pieces out of the plugin in the mid to long
term.
This makes running tests harder as we cannot call gst_init() after gst_deinit()
has been called.
This is what the API reference has to say about it at
https://gstreamer.freedesktop.org/documentation/gstreamer/gst.html?gi-language=c#gst_deinit
It is normally not needed to call this function in a normal application as the
resources will automatically be freed when the program terminates. This function
is therefore mostly used by testsuites and other memory profiling tools.
It isn't needed in the implementation either. It was only useful because it
included system headers like sys/types.h and sys/socket.h which we should now
include directly.
This will make it easier to move the media manager into the core sources.
Previously cui_call_get_display_name() would return the ID of the caller if no
contact was found. As this has changed recently the test for got_contact broke.
Shuffles some code around so that the property bindings are at the end.
This allows us to return early if there is a NULL contact (as is the case
for anonymous callers).
This makes sure that a NULL id will return a NULL contact and additionally
asserts that even a NULL contact will give us sensible primary/secondary display
information.
This allows to get rid of any special casing that the users of the
calls_best_match_get_name() and calls_best_match_get_id() had to do previously.
We also allow passing in NULL for *_get_primary_info ()
and *_get_secondary_info () for the anonymous caller case.
The designs for the call details show information on the type of the call:
https://gitlab.gnome.org/Teams/Design/app-mockups/blob/master/calls/calls.png
So f.e. "Cellular", "Matrix WebRTC Video Call".
These properties can potentially also be useful in choosing the mechanism to
use for the audio controls from the call display.
While not every class or interface is properly documented and annotated yet,
adding them to the docs should still be valuable.
Additionally it is a good reminder for ourselves to add some proper description
because things like missing a annotated class description really stand out
currently.
This was handled explicitly in the Call window.
By changing the logic to delay the emission of "ui-call-removed" we make sure
that the Call UI and the exported DBus object is consistent.
We also need to change the test cases to use run a GMainLoop because we now have
to wait until signal comes in.
