Replaced docusaurus with mkdocs

docs/guide-routr-as-asterisk-frontend.md

@@ -5,7 +5,7 @@ title: Routr as Asterisk Frontend
 
 This guide explores the use case of using Asterisk merely as a Media Server and a more specialized software, like **Routr**, to take care of the signaling and resource management. In other words, Asterisk will be in charge of the ivrs, voice mail, call recording, while **Routr** deals with connecting Agents, Peers, and Gateways. The following illustration depicts our scenario:
 
-<img src="../img/peering_ilustration.png" width=600 vspace=50>
+<img src="/assets/images/peering_ilustration.png" width=600 vspace=50>
 
 **Content**
 
@@ -47,22 +47,22 @@ The first file we will examine and change is `config/peers.yml`. Make note of th
 
 Head to the console and run the command `rctl -- get peers` to confirm that the Peer exist. The result should be as follows:
 
-<img src="../img/get_peers_cmd_output.png" width=600 >
+<img src="/assets/images/get_peers_cmd_output.png" width=600 >
 
 Next, we focus our attention to `domains.yml` and `agents.yml`. With a fresh installation, we don't need to make any changes to this files. However, you could run the commands `get domains` and `get agents` to ensure that both, the Agent and the Domain, exist on the server. Your output should look similar to:
 
-<img src="../img/get_domains_and_agents.png" width=600 >
+<img src="/assets/images/get_domains_and_agents.png" width=600 >
 
 Use the information in `agents.yml` to configure your SIP phone. The relevant information is found in `spec.credentials`. Mine looks like this:
 
-<img src="../img/john_telephone_setup_general.png" width=500 >
-<img src="../img/john_telephone_setup_advanced.png" width=500 >
+<img src="/assets/images/john_telephone_setup_general.png" width=500 >
+<img src="/assets/images/john_telephone_setup_advanced.png" width=500 >
 
 > Make the adjustments based on your prefer SIP phone.
 
 You can verify that your device registered correctly with **Routr** by running the `locate` command:
 
-<img src="../img/locate_john.png" width=600 >
+<img src="/assets/images/locate_john.png" width=600 >
 
 ## Configuring Asterisk
 
@@ -120,7 +120,7 @@ exten => 1001,n,Hangup
 
 Restart your Asterisk and check the location service. A new device will appear.
 
-<img src="../img/locate_john_and_ast.png" width=600 >
+<img src="/assets/images/locate_john_and_ast.png" width=600 >
 
 ## Calling Asterisk from John's device
 

docs/guide-securing-the-signal.md

@@ -7,7 +7,7 @@ Follow this guide to secure the signaling between your endpoints and **Routr**.
 
 > For this guide we will use a fictitious domain name to demonstrate the process of securing the signaling path
 
-<img src="../img/secure_signaling.png" width=600 vspace=30>
+<img src="/assets/images/secure_signaling.png" width=600 vspace=30>
 
 ## Creating a Java Keystore(.JKS) Certificate
 
@@ -94,8 +94,8 @@ openssl s_client -host 192.168.1.2 -port 5061    # Remember to use Routr's IP
 
 Go to the account that you want to secure, select `Advanced -> Sip Signaling` and change the parameter `Primary Proxy` to `${proxyHost}:${proxyPort};transport=tls`. See the example in the following image:
 
-<img src="../img/blinkpro_tls_config.png" width=600>
+<img src="/assets/images/blinkpro_tls_config.png" width=600>
 
 If everything went well you should see a green padlock like the one in the image bellow:
 
-<img src="../img/blinkpro_tls_secured.png" width=400>
+<img src="/assets/images/blinkpro_tls_secured.png" width=400>

docs/guide-voip-network-setup.md

@@ -121,26 +121,26 @@ Your output should be as follows:
 
 **Starting the Server**
 
-<img src="../img/starting_server.png" width=600 >
+<img src="/assets/images/starting_server.png" width=600 >
 
 **Verifying the Configuration**
 
-<img src="../img/verify_configuration.png" width=600 >
+<img src="/assets/images/verify_configuration.png" width=600 >
 
 ## Configuring the Sip devices
 
 > We are using "Telephone" for this example. You might use any softphone you wish, just keep in mind that the configuration will look slightly different.
 
 Configure your softphone using the information you gather in the last step. Start by completing only the required information: _username_, _domain_, _password_. Also, In the advanced section use the server's **IP** as your _Registry Server_ and _Proxy_. Here is how mine looks like:
 
-<img src="../img/telephone_config_general.png" width=500>
-<img src="../img/telephone_config_advanced.png" width=500>
+<img src="/assets/images/telephone_config_general.png" width=500>
+<img src="/assets/images/telephone_config_advanced.png" width=500>
 
 > Make sure to check the box "Use this account" to register your device
 
 If everything went well we just need to confirm that both softphones have registered correctly. Conveniently you can use the `.rctl locate` to obtain a list of "online" devices. This may seem like a lot of information. But what's relevant here is that both `1001` and `1002` are present in the location service and therefore can reach each other.
 
-<img src="../img/locate_agents.png" width=600>
+<img src="/assets/images/locate_agents.png" width=600>
 
 ## Making calls
 

docs/index.md

@@ -0,0 +1,45 @@
+---
+id: getting-started-introduction
+title: Undertanding Routr
+---
+
+**Routr** is a lightweight sip proxy, location server, and registrar that provides a reliable and scalable SIP infrastructure for telephony carriers, communication service providers, and integrators. It also provides with capabilities that are suitable for the enterprise and personal needs. To get involved in the development of this project please contact us at [@fonoster](https://twitter.com/fonoster).
+
+## Features
+
+- Proxy
+- Registrar Service
+- Location Service
+- Call Forking
+- Multi-Tenancy/Multi-Domain
+- Access to the PSTN Using SIP Gateways
+- Transport: TCP, UDP, TLS, WebSocket
+- Data Sources: Redis, Restful API, Files
+- Security
+  - Digest SIP User Authentication
+  - Domain Access Control List (DACL)
+  - RESTful service secured with TLS and JWT tokens
+- Rest API
+- Command Line Tool for Admin Operations
+- Routing Capabilities
+  - Intra-Domain Routing (IDR)
+  - Domain Ingress Routing(DIR)
+  - Domain Egress Routing (DER)
+  - Peer Egress Routing (PER)
+
+## Key Concepts
+
+This following table features some important concepts, including the different routing types implemented by the server.
+
+| Concept | Description |
+| -- | -- |
+| User    | Users perform administrative actions on the server              |
+| Agent   | Agents represent SIP endpoints such as softphones and IP phones |
+| Domain  | Enables the creation of isolated groups of Agents               |
+| Peer    | Similar to Agents but without Domain boundaries                 |
+| Gateway | SIP entity that allows call termination                         |
+| DID     | Routes and translate calls between the PSTN and Routr         |
+| Intra-Domain Routing  | Routing type for calling within the same Domain   |
+| Domain Ingress Routing | Calling from the PSTN to an Agent or Peer        |
+| Domain Egress Routing | Calling from an Agent to the PSTN thru a Gateway  |
+| Peer Egress Routing | Similar to *DER* but applies only to Peers          |