diff --git a/README.webrtc b/README.webrtc
index 260b7ba..25a5a2f 100644
--- a/README.webrtc
+++ b/README.webrtc
@@ -1,136 +1,161 @@
 
 SylkServer WebRTC gateway
 -------------------------
 
 The WebRTC gateway application enables web based clients to communicate
 transparently with SIP endpoints using WebRTC suite of standards and
 protocols.
 
 To perform this function SylkServer must run on a computer with a public
 reachable IP adddress.
 
 A ready to go client is available, see Sylk WebRTC client for more
 information.
 
 
 Architecture
 ------------
 
 +--------+               +-------------+ +----------+     +-------------+
 |        |               |             | |          |     |     SIP     |
 | WebRTC |  SylkRTC API  |             | |  Janus   +<--->+ audio/video |
 | client +<------------->+  SylkServer | |          |     |    calls    |
 |        |               |   WebRTC    | +----------+     +-------------+
-+--------+               |   gateway   | +----------+     +-------------+
-                         |             | | Presence |     |   SIMPLE    |
-                         |             | | and chat +<--->+     IM      |
-                         |             | | gateway  |     |   Presence  |
-                         +-------------+ +----------+     +-------------+
++--------+               |   gateway   | +----------+
+                         |             | | Chat and |
+                         |             | | file     |
+                         |             | | transfer |
+                         +-------------+ +----------+
 
 
 End-point requirements
 ----------------------
 
-The WebRTC gateway doesn't do media transcoding, so after taking care of the
+WebRTC gateway doesn't do media transcoding, so after taking care of the
 WebRTC-specific media features (ICE and DTLS mainly), endpoints will
 exchange plain RTP media between them, therefore SIP end-points must have a
 set of compatible audio (Opus) and video codecs (H.264 and VP8).
 
 As a WebRTC client, a standard browser with WebRTC support can be used
 (e.g.  Firefox or Google Chrome).
 
 
 Features
 --------
 
  * SIP account registration
- * Audio / video calling
+ * One-to-One Audio / video calling between SIP and Web clients
+ * Multi-party conferencing (WebRTC end-points only)
  * Codec agnostic
 
 
 SylkRTC API
 -----------
 
+Desktop API
+
 A JavaScript library implementing SylkRTC API is available at:
 
 https://github.com/AGProjects/sylkrtc.js
 
+Mobile API
+
+A React native API is available at:
+
+https://github.com/AGProjects/react-native-sylkrtc.js
+
+
+SIP Proxy
+---------
+
+A SIP Proxy is required for one-to-one calls.
+
+A guide for setting up a SIP Proxy is available here:
+
+https://sylkserver.com/download/#sip-proxy-setup
+
 
 Janus
 -----
 
 For implementing WebRTC capabilities, SylkServer uses an external component
 called Janus.
 
 https://github.com/meetecho/janus-gateway
 
 Janus can be installed by following the instructions on its README file:
 
 https://github.com/meetecho/janus-gateway/blob/master/README.md
 
 The only requirement is that Janus is compiled with WebSocket supports,
 since that's the transport SylkServer uses in order to communicate with
 Janus.
 
 To configure janus the following settings are relevant for WebRTC gateway
 (read their description in the janus configuration files):
 
 1. janus.cfg
 
 [general]
 api_secret = some_secret_key_shared_with_sylkserver
 ;debug_level = 5
 
 [nat]
 ice_lite = true
 ice_tcp = true
 ;ice_enforce_list = ip_or_interface
 ;ice_ignore_list = vmnet
 
 [media]
 force-bundle = true
 force-rtcp-mux = true
 
 [plugins]
 disable = libjanus_voicemail.so,libjanus_recordplay.so,libjanus_streaming.so,libjanus_echotest.so,libjanus_videocall.so,libjanus_textroom.so
 
 [transports]
 disable = libjanus_http.so,libjanus_rabbitmq.so,libjanus_pfunix.so
 
 2. janus.plugin.sip.cfg
 
 [general]
 ;local_ip = ip_for_sip
 keepalive_interval = 0
 user_agent = SylkServer (WebRTC gateway)
 register_ttl = 600
 ;rtp_port_range = 20000-40000
 
 3. janus.plugin.videoroom.cfg
 
 ; comment out the rooms defined in this file to prevent janus from
 ; creating some static rooms (having an empty file for it is fine)
 
 4. janus.transport.websockets.cfg
 
 [general]
 ws = yes
 ws_interface = lo
 ws_port = 8188
 wss = no
 
 [admin]
 admin_ws = no
 
 
 Running
 -------
 
 After installing SylkServer with WebRTC capability, start the server and
 point your web browser to:
 
 https://hostname:10888/webrtcgateway/
 
 It's recommended to change the API secret used between SylkServer and Janus,
 check the webrtcgateway.ini and janus.cfg configuration files.
 
+
+Configuration
+-------------
+
+See  webrtcgateway.ini.sample sample file for available options and their
+meaning.
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