check in callUri if not already in call
WEBRTC-50 : add test case
WEBRTC-51 : reenable auto-reconnect on non 503 scenarios
new ExSIP version
new ExSIP version
set margin of client .row to 0
add rigth and buttom attributes to localVideo selector
add top and left attributes to dialpad selector
escape empty spaces in sip userid uri
add style for audioOnly view to fix width
only start timer if not already started, use exsip.js for tests
change resolution css video height to use standard or widescreen and padding-bottom
fix muteAudio() updating Client.muted
add Client.setAudioOnly() and updating user media also when active session
224942 : handle CANCEL correctly
224942 : set status to confirmed on 200 INVITE response before setting remoteDescription, adding getDataSeriesByLabel()
pass createOfferConstraints to changeSession on updateUserMedia
set RTCConstraints to ua.rtcMediaHandlerOptions
change getStat to getStatAvg
refactor to handle multiple clients on the same page