diff --git a/README b/README
index bd2ea54..4f54ca5 100644
--- a/README
+++ b/README
@@ -1,106 +1,104 @@
 
 SylkServer
 ----------
 
 Copyright (c) 2010-2011 AG Projects
 http://ag-projects.com
 
 Authors: Adrian Georgescu, Denis Bilenko, Saul Ibarra
 Home page: http://sylkserver.com
 
 
 License
 -------
 
 SylkServer is licensed under GNU General Public License version 3. A copy of
 the license is available at http://www.fsf.org/licensing/licenses/agpl-3.0.html
 
 
 Description
 -----------
 
-SylkServer is a state of the art, extensible SIP Application Server
+SylkServer is an extensible SIP Application Server.
 
 SylkServer allows creation and delivery of rich multimedia applications
 accessed by SIP User Agents.  The server supports SIP signaling over TLS,
 TCP and UDP transports, RTP and MSRP media planes, has built in capabilities
-for creating ad-hoc SIP multimedia conferences with Wideband Audio, IM and
+for creating ad-hoc SIP multimedia conferences with wideband Audio, IM and
 File Transfer and can be easily extended with other applications by using
 Python programming language.
 
 
 Features
 --------
 
 SIP Signaling
 
  - TLS, TCP and UDP transports
- - INVITE and MESSAGE for sessions
- - SUBSCRIBE/NOTIFY
+ - INVITE and REFER methods
+ - SUBSCRIBE/NOTIFY mechanism
+ - Protocol tracing to file
+ - Outbound Proxy support
+ - Trusted peers
 
 Audio
 
- - Wideband codecs (G722 and Speex)
- - Narrow-band codecs (G711 and GSM)
+ - Codecs: G722, Speex, G711 and GSM
  - sRTP encryption
  - RTP timeout
- - DTMF handling
  
 Instant Messaging
 
- - MSRP chat and SIP MESSAGE
+ - MSRP chat
  - CPIM envelope
- - Is-composing indicator
- - History buffer
+ - Is-composing
 
 Conferencing
 
- - RTP wideband mixer
- - MSRP Switch with private messgaing
- - Conference event package
+ - Wideband RTP mixer
+ - MSRP switch
  - isfocus support
+ - Conference event package
+ - Add/remove participants
 
 
 Built-in Applications
 ---------------------
 
 1. Conference
 
 SylkServer allows SIP end-points to create ad-hoc conference rooms by
 sending INVITE to a random username at the hostname or domain where the
 server runs.  Other participants can then join by sending an INVITE to the
 same SIP URI used to create the room.  The INVITE and subsequent re-INVITE
 methods may contain one or more media types supported by the server.  Each
 conference room mixed audio, instant messages and uploded files are
 dispatched to all participants.
 
-SIP end-points that do not support MSRP chat can join the bridge by using
-audio only, they will receive the chat messages over the SIP signaling using
-SIP MESSAGE method, which is supported by many legacy end-points.  Messages
-sent to the room using SIP MESSAGE will be dispatched by either SIP MESSAGE
-or through established MSRP sessions depending on how the end-points have
-joined the room.
+One can remove or add participants by sending a REFER method to the
+conference URI.
 
 If a participant sends a file to the SIP URI of the room, the server will
 accept it, store it for the duration of the conference and offer it to all
 participants either present at that moment, or later to those that have
 joined the conference at a later moment.
 
 
 Standards
 ---------
 
 The server implements relevant features from the following standards:
 
  * MSRP and its relay extension RFC4975, RFC4976
  * MSRP switch draft-ietf-simple-chat-08
  * File Transfer over MSRP RFC5547
  * Indication of Message Composition RFC3994
  * CPIM Message Format RFC3862
  * Conference event package RFC4575
  * A Framework for Conferencing with SIP RFC4353
  * Conferencing for User Agents RFC4579
    5.1  INVITE: Joining a Conference Using the Conference URI - Dial-In
    5.2  INVITE: Adding a Participant by the Focus - Dial-Out
    5.5  REFER: Requesting a Focus to Add a New Resource to a Conference
    5.11 REFER with BYE: Requesting a Focus to Remove a Participant from a Conference
+