Upon accepting a video call on a device that can not establish a video track on
its own (for example by not having a camera), displaying the video enable/disable
button would fail. This commit defaults this button to disabled.
This is a feature of WebRTC that's [not standardized][1] and only
supported by libwebrtc. Since there's no support in jingle for passing
this capability from one peer to another, we're currently hard-coding
this option into both the local candidate and also the remote candidate
so they can use it.
But I'm trying to call a user that isn't using WebRTC, and renomination
is causing the call to stay in "connecting..." state for 10 or 20
seconds, sometimes longer, while both sides wait for the other to
nominate something based on their individual beliefs about the standards
they're using.
Removing this seems to make connecting relatively instantaneous.
If we want to reintroduce this feature, we should probably make a XEP so
the peers can negotiate honestly about it, and only use it if both sides
truely support the feature.
[1]: https://datatracker.ietf.org/doc/html/draft-thatcher-ice-renomination-01
Conversations would attempt to feed any candidates found in the session-accept into
WebRTC; even if the session wasn’t setup correctly.
this commit processes the candidates only if the session was setup correctly
fixes#3867
JingleRTPConnection shuts down the WebRTCWrapper before transitioning into a terminal state.
(This allows us to make sure it is actually closed when reaching that state);
However that means that, when we get a UI redrawn inbetween closing and transitioning we might
still be in SESSION_ACCEPTED but with no PeerConnection. This traditionally has triggered
an IllegalStateException on getting the EndUserState.
This commit catches the ISE and returns 'ENDING' instead.
Chances are that this is only visibiliy for a very brief time in the UI before the transition
triggers the UI to redraw with the proper state.
fixes#3848
some implementations will transform the following SDP coming from Firefox
m=audio 12346 RTP/AVP 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
to
<payload-type channels="1" name="telephone-event" clockrate="8000" id="101">
<parameter value="0-15" xmlns="urn:xmpp:jingle:apps:rtp:1"/>
</payload-type>
While a missing name attribute is not legal according to the XEP; and 0-15 are
technically not just one value the following commit will accept it if there is
just one paramater.
if the activity is not connected during finish it won’t receive the last end user state.
this code remembers it even if the actual session is already gone. so when activity reconnects and
we can’t find the real rtp session we can look up the last state instead.
