Merge pull request #13 from jevonearth/master

Add documentation.

README.md

@@ -1,45 +1,51 @@
 [![Build Status](https://drone.io/github.com/sippy/b2bua/status.png)](https://drone.io/github.com/sippy/b2bua/latest)
 
-# Sippy B2BUA
-Sippy B2BUA is a [RFC3261](https://www.ietf.org/rfc/rfc3261.txt)-compliant
-Session Initiation Protocol (SIP)
-(Back-to-back user agent)[http://en.wikipedia.org/wiki/Back-to-back_user_agent]i
-(B2BUA). 
+# Sippy B2BUA Sippy B2BUA is a
+[RFC3261](https://www.ietf.org/rfc/rfc3261.txt)-compliant Session Initiation Protocol (SIP) [Back-to-back user agent](http://en.wikipedia.org/wiki/Back-to-back_user_agent) (B2BUA). 
 
 The Sippy B2BUA is a SIP call controlling component. Unlike a SIP proxy server,
 which only maintains transaction state, the Sippy B2BUA maintains complete call
 state and participates in all call requests. For this reason it can perform
-number of functions that are not possible to implement using SIP proxy, such
-as for example accurate call accounting, pre-paid rating and billing, fail
-over call routing etc. Unlike PBX-type solutions such as Asterisk for example,
-the Sippy B2BUA doesn't perform any media relaying or processing, therefore it
-doesn't introduce any additional packet loss, delay or jitter into the
-media path. 
+number of functions that are not possible to implement using SIP proxy, such as
+for example accurate call accounting, pre-paid rating and billing, fail over
+call routing etc. Unlike PBX-type solutions such as Asterisk for example, the
+Sippy B2BUA doesn't perform any media relaying or processing, therefore it
+doesn't introduce any additional packet loss, delay or jitter into the media
+path. 
 
 ## Features
 
-- 5,000-10,000 simultaneous sessions per server;
-- 150-400 call setups/tear-downs per second;
-- Real-time calls control and call data records (CDRs) generation;
-- Optional ability to use (Sippy RTPproxy)[https://github.com/sippy/rtpproxy] for media relaying;
+- 5,000-10,000 simultaneous sessions per server
+- 150-400 call setups/tear-downs per second
+- Real-time calls control and call data records (CDRs) generation
+- Optional ability to use [Sippy RTPproxy](https://github.com/sippy/rtpproxy)
+  for media relaying
 - Optional ability to perform Cisco-compatible RADIUS AAA (Authentication,
-Authorization and Accounting);
-- RFC3261 compliance;
+  Authorization and Accounting)
+- RFC3261 compliance
 - Seamless compatibility with majority of popular SIP software and hardware on
-the market today;
-- Robustness and Resilience;
+  the market today
+- Robustness and Resilience
 - Simple and clean, yet flexible, internal design making implementing new
-features and services easy;
+  features and services easy
 - Sippy B2BUA could be easily combined with other Open Source software, such as
-SIP Express Router / OpenSIPS to build complete softswitch solution.
+  SIP Express Router / OpenSIPS to build complete softswitch solution
 
 ## Installation
 
 `pip install git+https://github.com/sippy/b2bua`
 
 ## Running
 
-To get started, you can use the b2bua_simple implementation. The following example will cause the b2bua run in the foreground so you can see the SIP messaging. If you make a call to the IP address of your host machine, the b2bua will recieve the call on its UAC side, and it will send a new call leg out its UAS side to the IP address 192.168.1.1. It is expected that 192.168.1.1 is some sort of SIP switch or termination gateway.
+To get started, you can use the b2bua_simple implementation. The following
+example will cause the b2bua run in the foreground so you can see the SIP
+messaging. If you make a call to the IP address of your host machine, the b2bua
+will recieve the call on its UAC side, and it will send a new call leg out its
+UAS side to the IP address 192.168.1.1. It is expected that 192.168.1.1 is some
+sort of SIP switch or termination gateway.
 
 `b2bua_simple -f -n 192.168.1.1`
 
+## Documentation
+
+See [documentation/documentation.md](documentation/documentation.md)