chore: add a lab for ha ivr with routr and asterisk

psanders · Feb 25, 2024 · 09f6e09 · 09f6e09
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diff --git a/README.md b/README.md
@@ -8,7 +8,7 @@ I'm just getting started with this, so there's only one lab at the moment.
 
 I'll be adding more soon.
 
-- [WebRTC Network with SIP.js and Routr](webrtc_network_with_sipjs_and_routr)
+- [WebRTC Network with SIP.js and Routr](webrtc_network_with_routr_and_sipjs)
 
 ## Support my work
 

diff --git a/calling_a_door_phone_from_sipjs/compose.yaml b/calling_a_door_phone_from_sipjs/compose.yaml
@@ -0,0 +1,36 @@
+version: "3.9"
+
+services:
+  routr:
+    image: fonoster/routr-one:latest
+    environment:
+      EXTERNAL_ADDRS: ${DOCKER_HOST_ADDRESS}
+      RTPENGINE_HOST: rtpengine
+    ports: 
+      - 51908:51908
+      - 5060:5060
+      - 5060:5060/udp
+      - 5062:5062
+    volumes:
+      - shared:/var/lib/postgresql/data
+
+  # RTPEngine requires of network mode "host" to work properly. However, this option doesn't work on 
+  # Windows and MacOs. For development, we are opening a few ports to the host machine. 
+  # For production, you must use the network_mode: host which works on Linux.
+  rtpengine:
+    image: fonoster/rtpengine:latest
+    restart: unless-stopped
+    # Uncomment the following line for production
+    # network_mode: host
+    environment:
+      # Set DOCKER_HOST_ADDRESS to an IP address that is reachable to the SIP clients
+      PUBLIC_IP: ${DOCKER_HOST_ADDRESS}
+      PORT_MIN: 10000
+      PORT_MAX: 10100
+      LOG_LEVEL: 6
+    ports:
+      - 22222:22222/udp
+      - 10000-10100:10000-10100/udp
+
+volumes:
+  shared:
diff --git a/calling_a_door_phone_from_sipjs/index.html b/calling_a_door_phone_from_sipjs/index.html
@@ -0,0 +1,70 @@
+<!DOCTYPE html>
+<html lang="en" xmlns="http://www.w3.org/1999/xhtml">
+
+<head>
+  <meta charset="utf-8" />
+  <title>Video Feed</title>
+  <style>
+    h1 { font-family: 'Courier New', Courier, monospace; }
+  </style>
+</head>
+
+<body>
+  <h1>Door Phone Video Feed</h1>
+  <div style="border: 1px solid black;  width: 300px; height: 300px;">
+    <video id="remoteVideo" autoplay muted style="background-color: black; height: 100%"></video>
+  </div>
+
+  <br />
+
+  <button id="callButton">Call Door Phone</button>
+  <button id="hangupButton">Hangup</button>
+
+  <audio style="display: none" id="remoteAudio" controls>
+    <p>Your browser doesn't support HTML5 audio.</p>
+  </audio>
+  <script type="module">
+    import { Web } from "https://unpkg.com/[email protected]/lib/index.js";
+
+    document.addEventListener('DOMContentLoaded', () => {
+      const remoteVideo = document.getElementById('remoteVideo');
+      const callButton = document.getElementById('callButton');
+      const hangupButton = document.getElementById('hangupButton');
+      const remoteAudio = document.getElementById('remoteAudio');
+
+      const config = {
+        server: "ws://localhost:5062",
+        aor:  "sip:[email protected]",
+        doorPhoneAor: "sip:[email protected]"
+      }
+
+      const options = {
+        aor: config.aor,
+        media: {
+          constraints: { audio: true, video: true },
+          remote: {
+            audio: remoteAudio,
+            video: remoteVideo
+          }
+        },
+        userAgentOptions: {
+          authorizationUsername: "admin",
+          authorizationPassword: "1234",
+        }
+      };
+
+      const simpleUser = new Web.SimpleUser(config.server, options);
+
+      callButton.addEventListener('click', async() => {
+        await simpleUser.connect();
+        simpleUser.call(config.doorPhoneAor);
+      });
+
+      hangupButton.addEventListener('click', () => {
+        simpleUser.hangup();
+      });
+    });
+  </script>
+</body>
+
+</html>
diff --git a/calling_the_pstn_from_sipjs/README.md b/calling_the_pstn_from_sipjs/README.md
@@ -0,0 +1,4 @@
+# Calling the PSTN from SIP.js
+
+This example shows how to use the SIP.js library to make a call from a web page to a phone number.
+
diff --git a/calling_the_pstn_from_sipjs/compose.yaml b/calling_the_pstn_from_sipjs/compose.yaml
@@ -0,0 +1,36 @@
+version: "3.9"
+
+services:
+  routr:
+    image: fonoster/routr-one:latest
+    environment:
+      EXTERNAL_ADDRS: ${DOCKER_HOST_ADDRESS}
+      RTPENGINE_HOST: rtpengine
+    ports: 
+      - 51908:51908
+      - 5060:5060
+      - 5060:5060/udp
+      - 5062:5062
+    volumes:
+      - shared:/var/lib/postgresql/data
+
+  # RTPEngine requires of network mode "host" to work properly. However, this option doesn't work on 
+  # Windows and MacOs. For development, we are opening a few ports to the host machine. 
+  # For production, you must use the network_mode: host which works on Linux.
+  rtpengine:
+    image: fonoster/rtpengine:latest
+    restart: unless-stopped
+    # Uncomment the following line for production
+    # network_mode: host
+    environment:
+      # Set DOCKER_HOST_ADDRESS to an IP address that is reachable to the SIP clients
+      PUBLIC_IP: ${DOCKER_HOST_ADDRESS}
+      PORT_MIN: 10000
+      PORT_MAX: 10100
+      LOG_LEVEL: 6
+    ports:
+      - 22222:22222/udp
+      - 10000-10100:10000-10100/udp
+
+volumes:
+  shared:
diff --git a/ha_ivr_with_routr_and_asterisk/README.md b/ha_ivr_with_routr_and_asterisk/README.md
@@ -0,0 +1,20 @@
+# Happy hacking!
+
+To start the service, use the following command:
+
+```bash
+# Remember to replace the Docker Host Address with your own
+DOCKER_HOST_ADDRESS=192.168.1.3 docker compose up
+```
+
+> Wait for all the services to be "healthy". 
+
+Then, you will need to register with Routr from a softphone like Zoiper.
+
+The access information is as follows:
+
+Domain: sip.local
+Username: 1001
+Password: 1234
+
+After this, you can make a call to Asterisk using sip:asterisk@routr as the address of record.
diff --git a/ha_ivr_with_routr_and_asterisk/compose.yaml b/ha_ivr_with_routr_and_asterisk/compose.yaml
@@ -0,0 +1,58 @@
+version: "3"
+
+services:
+  routr:
+    image: fonoster/routr-one:2.7.0
+    environment:
+      EXTERNAL_ADDRS: ${DOCKER_HOST_ADDRESS}
+      RTPENGINE_HOST: rtpengine
+    ports:
+      - 51908:51908
+      - 5060:5060
+      - 5062:5062
+    volumes:
+      - shared:/var/lib/postgresql/data
+
+  rtpengine:
+    image: fonoster/rtpengine:latest
+    ports:
+      - 22222:22222/udp
+      - 10000-10100:10000-10100/udp
+    environment:
+      PUBLIC_IP: ${DOCKER_HOST_ADDRESS} 
+      PORT_MIN: 10000
+      PORT_MAX: 10100
+
+  asterisk-01:
+    image: fonoster/mediaserver:latest
+    expose:
+      - 6060
+    environment:
+      EXTERN_ADDR: ${DOCKER_HOST_ADDRESS}
+      SIPPROXY_HOST: routr
+      SIPPROXY_USERNAME: asterisk
+      SIPPROXY_SECRET: asterisk
+    volumes:
+      - ./extensions.conf:/etc/asterisk/extensions.conf
+      - ./sounds/count-1.sln16:/var/lib/asterisk/sounds/count.sln16
+
+  asterisk-02:
+    image: fonoster/mediaserver:latest
+    expose:
+      - 6060
+    environment:
+      EXTERN_ADDR: ${DOCKER_HOST_ADDRESS}
+      SIPPROXY_HOST: routr
+      SIPPROXY_USERNAME: asterisk
+      SIPPROXY_SECRET: asterisk
+    volumes:
+      - ./extensions.conf:/etc/asterisk/extensions.conf
+      - ./sounds/count-2.sln16:/var/lib/asterisk/sounds/count.sln16
+
+  simplephone:
+    image: psanders/simplephone:latest
+    ports:
+      - 8080:8080
+
+volumes:
+  shared:
diff --git a/ha_ivr_with_routr_and_asterisk/extensions.conf b/ha_ivr_with_routr_and_asterisk/extensions.conf
@@ -0,0 +1,4 @@
+[local-ctx]
+exten => asterisk,1,NoOp()
+ same => n,Playback(count)
+ same => n,Hangup()
diff --git a/ha_ivr_with_routr_and_asterisk/sounds/count1.sln16 b/ha_ivr_with_routr_and_asterisk/sounds/count1.sln16
diff --git a/ha_ivr_with_routr_and_asterisk/sounds/count2.sln16 b/ha_ivr_with_routr_and_asterisk/sounds/count2.sln16
diff --git a/...tc_network_with_sipjs_and_routr/README.md → ...tc_network_with_routr_and_sipjs/README.md b/...tc_network_with_sipjs_and_routr/README.md → ...tc_network_with_routr_and_sipjs/README.md
diff --git a/...network_with_sipjs_and_routr/compose.yaml → ...network_with_routr_and_sipjs/compose.yaml b/...network_with_sipjs_and_routr/compose.yaml → ...network_with_routr_and_sipjs/compose.yaml
diff --git a/...c_network_with_sipjs_and_routr/etc/db.sql → ...c_network_with_routr_and_sipjs/etc/db.sql b/...c_network_with_sipjs_and_routr/etc/db.sql → ...c_network_with_routr_and_sipjs/etc/db.sql
diff --git a/...work_with_sipjs_and_routr/etc/pg_hba.conf → ...work_with_routr_and_sipjs/etc/pg_hba.conf b/...work_with_sipjs_and_routr/etc/pg_hba.conf → ...work_with_routr_and_sipjs/etc/pg_hba.conf
Original file line number	Diff line number	Diff line change
		@@ -0,0 +1,4 @@
		# Calling the PSTN from SIP.js

		This example shows how to use the SIP.js library to make a call from a web page to a phone number.