jiffyphone [5]: comment out hold code that breaks FF accepting new calls
PaaS [53]: sendDTMF -> dtmf()
Session cleanup
Spec Helper for fake WebRTC objects
Asynchronous test work and disabled a ton of tests
Session: getLocalStreams/getRemoteStreams return [] when peerConnection is closed
switched max_forwards from 69 to 70; more test cleanup and async additions
more UA tests, stability improved
Don't crash things on double-terminate (bad for refers)
more tests; session change to deal with 200 OK's without bodies client side
Does not send SDP in the 200 if it already set it up in reliable early media
emit 'connecting' from all gUM callbacks
finished UA testing, some config checks need reworking'
client no longer PRACKS retransmitted 183's; server will reply 200 if this occurs, as it should
Session: remove warning message from get*Streams when peerConnection is closed
Stop localMedia before following REFER (chrome bug)
added autostart parameter, if autostart === true then the user agent starts immediately
[79] refactored cancel method into switch statement, ISC receiveRequest
Session.js: InviteServerContext() checks for request.body before editing it
Enhancement #78. Allow SIPS scheme in remote URIs
Cherry-picked from JsSIP's 'develop' branch, then amended with bugfixes.
See https://github.com/versatica/JsSIP/commit/c3359b300e037a73a8d3894faabffbd956ff88dd

- Do not reject messages comming with SIPS scheme in remote URIs
- Reject incoming requests targeting SIPS scheme with a "416 Unsupported URI Scheme"