Now you can check the "Remote offerer" checkbox in the JS example to
force the peer to send the SDP offer. This involved implementing
support for receiving the OFFER_REQUEST message in the C example.
As a side-effect of this, the C example will no longer send
OFFER_REQUEST automatically when the --our-id option is passed. It
will only do so when the --remote-offerer option is explicitly passed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/31>
Pragmatically, an answer cannot be created until the offer is created as
the answer creation needs information from the offer. Practically, due
to implementation details, the answer was always queued after the set of
the offer and so the call flow did not matter.
The current code also hid a bug in webrtcbin where ice candidates would be
generated before the answer had been created which is against the JSEP
specification.
Change to the correct call flow for exemplary effect.
Add a switch to the command line utility that makes it request
the initial offer from the peer instead of generating it.
Modify the webrtc.js example to support a new REQUEST_OFFER
message, and generate the offer when receiving it.
We don't own the reference. Since GLib 2.58, the g_bytes_unref that
follows the signal emission in libsoup loudly complains about the
attempt to underflow the refcount.
This reduces the chance that someone will try to change the
audio/video source elements and get an error because they don't know
about the conversion elements. They will be no-ops in the usual case.
Closes https://github.com/centricular/gstwebrtc-demos/issues/8
When using localhost signalling servers, we don't want to use
strict ssl because it's probably using a self-signed certificate
and there's no need to do certificate checking over localhost anyway.
