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.
This should be good enough to survive some network switches where both networks are online at the same time to allow for some handover
(for example when enabling wifi the 3G connection will usually (probably depends on OS) live on for a moment
turns out libwebrtc doesn’t use the system root CA store but comes with only a few default CAs.
in anyway we will probably only use tcp/443 to bypass firewalls and not to actually secure anything.
