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.
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 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.
Introduce a state-changed signal which also gives a reason for why the state
changed. This will allow the UI to give some meaningful feedback to the user.
Additionally we can get rid of a number of things that were not really states,
but rather reasons for why a state changed (f.e. authentication failures).
Sofia detects a NAT by presence of the "received" parameter in the Via header in
the response to a REGISTER. Sofia will then update the Contact header to use the
IP as reported by the registrar.
The "received" parameter MUST be included in the response according to
https://datatracker.ietf.org/doc/html/rfc3261#section-18.2.1
when the registrar detects a difference between the domain part of the top Via
header and the packet source address but practice has shown that this will not
always be the case.
Addditionally this change allows us to have origins bound to different network
interfaces which would be useful when a registrar can only be accessed through a
VPN.
This also fixes an issue with SDP introduced in
36880c3d34 which was only seen on some SIP
providers:
The session name ("s=") line is not relevant for establishing a connection,
the connection data (c=") line is.
See https://datatracker.ietf.org/doc/html/rfc4566 section 5.3 and 5.7
Since we cannot do encrypted media streams yet, we should hardcode whether or
not we want to use SRTP to FALSE, so that sips target URLs can be used in SIP
calls at all.
If the origin is used for PSTN telephony extract the number from the
SIP dialstring (i.e. sip:+49160123456789@my-sip-host.de) and pass that
to call object for contact matching.
This let's us get rid of a lot of duplication in the derived classes.
Additionally we set the initial state to CALLS_CALL_STATE_INCOMING if
inbound is TRUE and CALLS_CALL_STATE_DIALING otherwise.
In this case network changes will not be detected.
Additionally fall back to binding on all network interfaces (in this case a user
will have problems when using multiple network interfaces, but there is really
not much we can do without a functioning CallsNetworkWatch).
Otherwise we might miss the IP of the remote peer leaving us unable to
establish a connection for RTP.
From https://datatracker.ietf.org/doc/html/rfc4566#section-5.7
A session description MUST contain either at least one "c=" field in
each media description or a single "c=" field at the session level.
It MAY contain a single session-level "c=" field and additional "c="
field(s) per media description, in which case the per-media values
override the session-level settings for the respective media.
The assumption that the IP of the remote peer can always be found in the
sdp_connection member of the sdp_session_s struct does not always hold true
and we should handle this case gracefully (i.e. without crashing).
As it's not guaranteed that the home directory is always writable
during the build. Debspawn for example does not allow this
and we might get such a warning:
`CallsSipProvider-WARNING **: 21:58:14.839: Failed to create directory '/home/salsaci/.config/calls': 13`
This makes sure all of the supported protocols have a chance of working.
Since nua_set_params does not update NUTAG_URL (carefully rechecking the docs
verifies this), it is safe to remove the code in update_nua().
However, this means that we will have to recreate the nua stack,
which incidentally is currently being worked on:
https://gitlab.gnome.org/GNOME/calls/-/merge_requests/402
secret_password_*_finish() may return FALSE without setting the GError.
F.e. trying to remove a non existent secret is not a failure.
The bug supposedly manifests itself because the updating account credentials
from the UI does not always seem to work correctly.
