diff --git a/README.webrtc b/README.webrtc
index fae4f49..260b7ba 100644
--- a/README.webrtc
+++ b/README.webrtc
@@ -1,138 +1,136 @@
 
 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  |
                          +-------------+ +----------+     +-------------+
 
 
 End-point requirements
 ----------------------
 
 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
  * Codec agnostic
 
 
 SylkRTC API
 -----------
 
 A JavaScript library implementing SylkRTC API is available at:
 
 https://github.com/AGProjects/sylkrtc.js
 
 
 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.
 
-Note: libsrtp >= 1.5.0 is recommended.
-
 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.