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Due to issues with the upstream release scripts (containing non-free/
pre-compiled tools/artifacts), this package is built from a repackaged
upstream tarball.  The repackaged upstream tarball is tracked in
github:

  https://github.com/opentelecoms-org/sipml5

The package libjs-sipml5 provide a JavaScript library for use in web sites
that want to offer WebRTC audio/video calling capabilities or any
other interactive service based on SIP over WebSockets as well.

The related sipml5-web-phone package demonstrates how to use this
library.

The library could be integrated into almost any web application,
include a static web site, a CMS or a Java web application.

It doesn't just work with any SIP server/proxy: it requires a SIP proxy
that supports the SIP over WebSockets transport.  This is currently
under development in most leading SIP proxies, for example, see
the b-webrtc branch of reSIProcate/repro:

  http://www.resiprocate.org/WebRTC_and_SIP_Over_WebSockets

These implementations may not be available in official Debian packages
just yet, but they can be easily built from source on a Debian system.

Furthermore, users accessing the site require a WebRTC capable browser:

  Firefox nightly build    
      http://nightly.mozilla.org/

  Chrome 25 or later       
      https://www.google.com/intl/en/chrome/browser/beta.html

WebRTC requires a TURN server.  There are two TURN servers available
in Debian:

  reTurn from reSIProcate:
      http://packages.debian.org/wheezy/resiprocate-turn-server

  Open TurnServer.org:
      http://packages.debian.org/sid/turnserver

Finally, the WebRTC browser/phone may insist on some of the following:

  SRTP: any device you call must also support SRTP

  AVPF (SAVPF): many standard SIP devices just support regular AVP.

  Codecs: Opus and G.711 are the core codecs for WebRTC.  Your browser
  may support others.  Most deskphones support G.711, but not Opus.