diff --git a/.env.example b/.env.example
index 28515ce..53f16e6 100644
--- a/.env.example
+++ b/.env.example
@@ -36,3 +36,7 @@ MONGODB_PORT_NUMBER=27017
 # Nats
 NATS_PORT_NUMBER=4222
 NATS_BIND_IP=127.0.0.1
+
+# SIP
+VOIP_DOMAIN=valid.voip.domain.com
+FREESWITCH_PASSWORD=admin_1234
diff --git a/README.md b/README.md
index a314fc7..d06f155 100644
--- a/README.md
+++ b/README.md
@@ -136,4 +136,16 @@ When running multiple Rocket.Chat servers, you can configure Traefik to discover
 ```env
 ROCKETCHAT_BACKEND_SERVERS=rocketchat-1:3000,rocketchat-2:3000,rocketchat-3:3000
 ```
-
+### VoIP setup
+There is a quick way for rolling out a VoIP solution using Drachtio and FreeSwitch as a SIP proxy. To achieve this some network configurations are required. `compose.network.yml` defines a network private network called `rc_voip_private` for VoIP servers. Currently it uses subnet 172.28.0.0/16 and assigns static IP 172.28.0.10 and 172.28.0.11. If you would want to change this network configuration, add desired value for subnet and static ip addresses in that subnet to `freeswitch` and `drachtio` service. `compose.network.yml` also configures rocketchat service to have access to default and `rc_voip_private` network. It communicates with non VoIP services on default network and VoIP services on `rc_voip_private` network. Once the desired configuration changes are made, run
+```bash
+    docker compose \
+        -f compose.monitoring.yml \
+        -f compose.traefik.yml \
+        -f compose.database.yml \
+        -f compose.yml \
+        -f compose.voip.yml \
+        -f compose.network.yml \
+        up -d
+```
+to start the stack. Once the stack is running, do the necessary VoIP configurations using the static ips for FreeSwitch and Drachtio.
diff --git a/compose.voip.yml b/compose.voip.yml
new file mode 100644
index 0000000..9116840
--- /dev/null
+++ b/compose.voip.yml
@@ -0,0 +1,29 @@
+services:
+  freeswitch:
+    image: docker.io/rocketchat/freeswitch:latest
+    # build: .
+    restart: always
+    environment:
+      EXTERNAL_ADDRESS_FQDN: ${VOIP_DOMAIN:-}
+      FREESWITCH_PASSWORD: ${FREESWITCH_PASSWORD:-}
+      RTP_START_PORT: 20000
+      RTP_END_PORT: 20046
+      DEBUG_MODE: "true"
+      ACCEPT_EXTERNAL_CALLS: false
+      USE_LEGACY_ARCH: false
+      DRACHTIO_DOMAIN: ${VOIP_DRACHTIO_DOMAIN:-drachtio}
+    expose:
+      - 5060
+    ports:
+      - 20000-20046:20000-20046/udp
+
+  drachtio:
+    image: drachtio/drachtio-server:latest
+    restart: always
+    ports:
+      - "9022:9022"
+      - "5060:5060/udp"
+    volumes:
+      # Drachtio params are configured on this file
+      # You must modify it to configure a strong and unique secret string
+      - ./files/drachtio/drachtio.conf.xml:/etc/drachtio.conf.xml
diff --git a/files/drachtio/drachtio.conf.xml b/files/drachtio/drachtio.conf.xml
new file mode 100644
index 0000000..7d88ec4
--- /dev/null
+++ b/files/drachtio/drachtio.conf.xml
@@ -0,0 +1,200 @@
+<drachtio>
+
+    <!-- udp port to listen on for client connections (default 9022), and shared secret used to authenticate clients -->
+	<admin port="9022" secret="set a strong secret here">0.0.0.0</admin>
+
+    <!-- the server can either accept inbound connections from node apps, or make outbound requests
+        to node apps.  You must globally choose one or the other approach but not both.
+
+        Inbound connections requires no configuration other than the <admin/> tag above, and are useful
+        when a server is hosting a single application, or has one primary function.
+
+        If, on the other hand, you want a server to support multiple applications, then we have the issue
+        of determining how to route an incoming request (INVITE, REGISTER, etc) to the appropriate application.
+        This is what outbound connections are designed for. 
+
+        By uncommenting the <request-handlers/> configuration below, you are placing the server into outbound mode, and 
+        providing an HTTP URL that the server will hit to determine how to route an incoming request.
+        The following parameters will be passed to the http service as query string parameters in an HTTP GET:
+
+            callingNumber - the user part of the From header
+            calledNumber - the user part of the To header
+            contentType - the Content-Type header (if any)
+            domain - the domain in the request uri
+            method - the sip method
+            protocol - the transport protocol
+            source_address - the sending address
+            uriUser - the user part of the sip uri
+            uri - the complete sip uri
+
+        It is the responsibility of the HTTP server that receives the request to return a json response 
+        with a top-level "action" of either "reject", "redirect", "proxy", or "route".
+
+            reject - reject the call with a specified status code
+            redirect = redirect the call to one or more specified sip uris
+            proxy - proxy the call to one or more specified sip uris
+            route - connect to a nodejs application that is listening on the specified host:port
+
+            example responses:
+
+            {
+                "action": "reject":
+                "data": {
+                    "status": 503,
+                    "reason": "Max Calls Exceeded"
+                }
+            }
+
+            {
+                "action": "redirect":
+                "data": {
+                    "contacts": [
+                        "sip:foo@bar.com"
+                    ]
+                }
+            }
+
+            {
+                "action": "proxy":
+                "data": {
+                    "destination": [
+                        "sip:foo@bar.com"
+                    ],
+                    recordRoute: true
+                }
+            }
+
+            {
+                "action": "route":
+                "data": {
+                    "url": "call-recording.default.svc.cluster.local:4000"
+                }
+            }
+
+            {
+                "action": "route":
+                "data": {
+                    "tag": "my-conferencing-app"
+                }
+            }
+
+    -->
+
+    <!-- comment this in and edit http url to use outbound connections
+         Note: currently only HTTP GET is supported as an HTTP METHOD
+    <request-handlers>
+        <request-handler sip-method="INVITE" http-method="GET">http://35.187.89.96:80</request-handler>
+    </request-handlers>
+    -->
+    <!-- sip configuration -->
+    <sip>
+        <contacts>
+            <!-- <contact>sip:*;transport=tcp</contact> -->
+            <contact>sip:*;transport=udp</contact>
+            <!-- local sip address to bind to.  Default: 'sip:*', which means listens on port 5060 on all interfaces and transports -->
+            <!--
+                    Other examples:
+                        sip:192.168.1.100        listen on a specific address
+                        sip:*;transport=tcp      tcp connections only
+                        sip:*:5061               listen on port 5061
+            -->
+
+
+            <!-- here's how to listen for every protocol on their default ports -->
+<!--
+            <contact>sip:*:80;transport=ws</contact>
+            <contact>sips:*;transport=tls</contact>
+            <contact>sips:*:443;transport=wss</contact>
+-->
+
+        </contacts>
+
+        <!-- uncomment and modify these if you want to change the sip stack timers (globally)
+            <timers>
+                <t1>500</t1>
+                <t2>4000</t2>
+                <t4>5000</t4>
+                <t1x64>32000</t1x64>
+            </timers>
+        -->
+
+        <!-- uncomment this if you send encapsulated SIP messages to homer
+        <capture-server port="9060" hep-version="3" id="101">127.0.0.1</capture-server>
+        -->
+
+        <!-- if you want to terminate SIP over TLS connections (example shows letsencrypt.org typical file locations, but use any CA)
+        <tls>
+                <key-file>/etc/letsencrypt/live/yourdomain/privkey.pem</key-file>
+                <cert-file>/etc/letsencrypt/live/yourdomain/cert.pem</cert-file>
+                <chain-file>/etc/letsencrypt/live/yourdomain/chain.pem</chain-file>
+                <dh-param>/var/local/dh4096.pem</dh-param>
+        </tls>
+        -->
+
+        <!-- if you want to increase the default mtu size for udp packets; stack will force tcp when sending larger packets
+        <mtu-size>4096</mtu-size>
+        -->
+        <udp-mtu>8192</udp-mtu>
+
+        <!-- uncommenting this will cause all outbound new requests (i.e. requests outside of a dialog) to go through the specified proxy -->
+        <!--
+        <outbound-proxy>sip:10.10.10.1</outbound-proxy>
+        -->
+
+        <!-- optional list of header values which, if encountered in a message will trigger an action
+             [tcp-]action="reject|discard"
+             reject means send a sip non-success response immediately
+             discard means sliently discard the request, generating no response
+        -->
+        <spammers action="reject" tcp-action="discard">
+            <header name="User-Agent">
+                <value>sip-cli</value>
+                <value>sipcli</value>
+                <value>friendly-scanner</value>
+            </header>
+            <header name="To">
+                <value>sipvicious</value>
+            </header>
+        </spammers>
+    </sip>
+
+    <!-- set to true if you want the server to cdr events to a connected client -->
+    <cdrs>false</cdrs>
+
+    <monitoring>
+        <prometheus port="8088">127.0.0.1</prometheus>
+    </monitoring>
+
+    <!-- logging configuration -->
+    <logging>
+
+        <!-- uncomment this if you want logging dumped to console, in addition to any other log destinations
+    -->
+        <console/>
+
+        <!-- uncomment if you want logging sent to syslog (over UDP) -->
+        <!--
+        <syslog>
+            <address>my.syslog.serve</address>
+            <port>514</port>
+            <facility>local6</facility>
+        </syslog>
+        -->
+
+        <!-- file-based logging -->
+        <file>
+            <name>/var/log/drachtio/drachtio.log</name>
+            <archive>/var/log/drachtio/archive</archive>
+            <size>100</size>
+            <maxFiles>20</maxFiles>
+            <auto-flush>true</auto-flush>
+        </file>
+
+        <!-- sofia (internal sip library) log level, from 0 (minimal) to 9 (verbose) -->
+        <sofia-loglevel>3</sofia-loglevel>
+
+        <!-- general process log level: notice, warning, error, info, debug.  Default: info -->
+        <loglevel>info</loglevel>
+    </logging>
+
+</drachtio>
\ No newline at end of file