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.
