Update WebRTC SIP SDK for web

Please download our latest update of WebRTC SIP SDK.

Change log:
– outgoing call connection timeout reduced
– custom headers for invite messages made possible
– customizable user-agent SIP field added
– DTMF implemented
– possibility to perform multiple simultaneous calls added
– video/audio recording implemented
– application interface changes (done more consistent, obsolete methods removed, parameters changed, renamings)
– possibility to select camera and microphone added
– tag added to TO SIP header field
– customizable register expiration timeout
– some of the deprecation warnings resolved
– hold redone to be more flexible with different servers
– various fixes
– new developer manual written