SIPrec-based recording using a drachtio server, and FreeSWITCH as the media server

This document provides a comprehensive guide to setting up SIPREC-based call recording based on Drachtio and using FreeSWITCH as the backend recording system. Drachtio, an open-source SIP server framework, enhances the implementation of SIPREC by providing a flexible, scalable, and programmable environment to handle SIP signaling. It acts as a bridge between the SIPREC clients and recording servers like FreeSWITCH.

Requirements

Software Requirements

Item	Recommended
Operating system	Debian 10
CPUs	4
RAM	8 GB
Storage	100 GB
Backend-recording system	FreeSWITCh
Runtime-environment	Node JS

Port Utilization Requirements

The following ports must be open on the server.

Type	Application	Description	Port
TCP	FreeSwitch	ESL port	8021
TCP	FreeSwitch	Websocket port	7443
TCP & UDP	FreeSwitch	Internal SIP UAS	5065
TCP & UDP	FreeSwitch	External SIP UAS	5080
TCP	FreeSwitch	Used for webRTC	5066
TCP & UDP	FreeSwitch	For NAT profile	5070
UDP	FreeSwitch	H.323 gatekeeper RAS	1719
TCP	FreeSwitch	H.323 call signaling	1720
TCP	FreeSwitch	MSRP(used for calls with messaging)	2855-2856
UDP	FreeSwitch	STUN service, for NAT traversal	3478-3479
TCP	FreeSwitch	MLP protocol server	5002
UDP	FreeSwitch	Neighborhood service	5003
UDP	FreeSwitch	RTP/ RTCP multimedia streaming	16384-32768
TCP & UDP	Drachtio	For incoming SIPrec	5060

The ports can be opened as follows:

Run the following command and replace PORT with each of the required ports listed above:

CODE

sudo iptables -A INPUT -p tcp -m tcp --dport PORT-j ACCEPT

Example:

CODE

sudo iptables -A INPUT -p tcp -m tcp --dport 8021 -j ACCEPT

Save this port configuration with the command:
- CODE
```
sudo iptables-save
```

Setting up the drachtio server

Drachtio-server is a SIP server that is built on the Sofia SIP stack. It provides a high-performance SIP engine that can be controlled by client applications written in pure Javascript running on node.js.

The node.js module that will be used to create applications controlling the server is called drachtio-srf. Install the following dependencies

CODE

sudo apt install libcurl4-openssl-dev

After installing libcurl, do as follows:

CODE

git clone --depth=50 --branch=main https://github.com/drachtio/drachtio-server.git && cd drachtio-server
git submodule update --init --recursive
./autogen.sh
mkdir build && cd $_
../configure CPPFLAGS='-DNDEBUG'
make
sudo make install

After the successful installation move the drachtio.conf.xml file to /etc

CODE

mv drachtio-server/drachtio.conf.xml /etc/

Open this file

CODE

nano /etc/drachtio.conf.xml

Replace 127.0.0.1 with the IP of your server. It’ll look like this

CODE

<admin port="9022" secret="cymru">192.168.1.90</admin>

To run the drachtio server do

CODE

sudo drachtio --config /etc/drachtio.conf.xml

Now make a service for the drachtio server

CODE

sudo nano /etc/systemd/system/drachtio.service

Insert the following line in this file

CODE

[Unit]
Description=Drachtio Server
After=network.target

[Service]
Type=simple
ExecStart=/usr/local/bin/drachtio
Restart=always

[Install]
WantedBy=multi-user.target

To start drachtio as a service

CODE

sudo systemctl daemon-reload
sudo systemctl enable drachtio
sudo systemctl start drachtio

Redis set up

Redis is used in this application to store the SIPrec invite and generate SDP for legB.

Install and start Redis.

CODE

sudo apt install redis
sudo systemctl start redis.service
sudo systemctl status redis.service

Freeswitch configuration

Configure freeswitch to record the calls coming from SBC/CUBE.

Make sure that the internal SIP profile is listening on 5065 as the 5060 port is used by drachtio for listening SIPrec from CUBE.

Now open public dialplan

CODE

nano /etc/freeswitch/dialplan/public.xml

Add the following dialpan to it

CODE

  <extension name="hairpin_and_record">
    <condition field="${sip_h_X-Return-Token}" expression="^(.+)$">
      <action application="export" data="sip_h_X-Return-Token=${sip_h_X-Return-Token}" />
      <action application="export" data="_nolocal_jitterbuffer_msec=100"/>
      <action application="set" data="RECORD_STEREO=true"/>
      <action application="set" data="call_id=${strftime(%Y%m%d_%H%M%S)}_${sip_from_tag}"/>
      <action application="set" data="outfile=$${base_dir}/recordings/${call_id}.wav"/> 
      <action application="record_session" data="${outfile}"/>
      <action application="set" data="hangup_after_bridge=true"/> 
      <action application="bridge" data="sofia/external/${destination_number}@${network_addr}"/>
    </condition>
  </extension>

Create directory recordings in your base directory.

Node.js application set up

This application requires a drachtio SIP server to be installed in your network.

Now clone this application

CODE

git clone  https://github.com/drachtio/drachtio-siprec-recording-server.git

Now enter the cloned directory and do

CODE

cp config/default.json.example-freeswitch config/local.json

Make config directory if node available.

Now open local.json and provide drachtio host (server IP), drachtio port (9022), freeswitch (server IP:5065), redis host(123.0.0.1), and redis port (6379).

We need to update the javascript code.

Open the lib/freeswitch-call-handler.js file and replace the createSdpForResponse() function with this

CODE

async function createSdpForResponse(opts, sdp, res) {
  try {
    await client.connect(); // Ensure the client is connected

    const result = await client.get(opts.sessionId);

    const combinedPayload = payloadCombiner(sdp, result, opts.sdp1, opts.sdp2);
    console.log('----------------------------- Created SDP for response.')
    return combinedPayload; // Resolve the promise with the combined payload
  } catch (err) {
    console.error('---------------Error in createSdpForResponse:', err);
    throw err; // Reject the promise with the error
  } finally {
    await client.quit(); // Ensure the client is properly closed
  }
}

Replace the storeUnusedSdp() function with this

CODE

async function storeUnusedSdp(opts) {
  try {
    debug(`sessionId: ${opts.sessionId}: sdp ${opts.sdp2}`);
    await client.connect();  // Ensure the client is connected

    // Using the promise-based version of set
    const reply = await client.set(opts.sessionId, opts.sdp2, 'EX', 10);
    
    console.log('Store the unused SDP in redis     -----------------------  ' + reply);

    return opts;  // Resolving the promise with opts
  } catch (err) {
    console.error('Could not store the unused SDP in redis:', err);
    throw err;  // Rejecting the promise with the error
  } finally {
    await client.quit();  // Ensure the client is properly closed
  }
}

Replace the exchangeSdp() function with this

CODE

async function exchangeSdp(sessionId, sdp) {
  try {
    await client.connect(); // Ensure the client is connected

    const replies = await client.multi()
      .get(sessionId)
      .set(sessionId, sdp)
      .exec();

    return replies[0]; // Resolve the promise with the result of the GET command
  } catch (err) {
    console.error('----------------- Error in exchangeSdp:', err);
    throw err; // Reject the promise with the error
  } finally {
    await client.quit(); // Ensure the client is properly closed
  }
}

Now save this file and run the following commands in drachtio-siprec-recording-server directory to run app.js

CODE

npm install drachtio
npm install redis config
npm init
npm cache clean --force
npm install -g npm@latest
npm install -g npm@10.8.1
node app.js

Testing

The tests done and their results are as below,

Test	Result
Normal call	Working
Consult call	Not working, Just the audio of the first session (A1 <--> C) before consulting is recorded, no audio recording (silent recording) for the second session (A1 <--> A2), and when the consult call is retrieved the voice (A1 <--> C) starts getting recording again.
Consult-Transfer call	Not working, Just the audio of the first session (A1 <--> C) before consulting is recorded, no audio recording (silent recording) for the second session (A1 <--> A2), and when the consult call is transferred the voice (A2 <--> C) still doesn’t,t get recorded.
Direct-Transfer call	Working, The audio of the first session (A1 <--> C) before consulting is recorded, and audio recording continues when the call is direct-transferred (C <-->A2)n.
Hold- Unhold call	The MOH is also recorded in the audio file
Pause-Resume call recording	This feature is working for SIPrec recording ( Performing it by hardcoding the recording file name)

Optional: Freeswitch configuration for pause-resume recording.

To add pause-resume recording update the dialplan inserted earlier with this

CODE

<extension name="hairpin_and_record">
    <condition field="${sip_h_X-Return-Token}" expression="^(.+)$">
      <action application="export" data="sip_h_X-Return-Token=${sip_h_X-Return-$      
      <action application="export" data="_nolocal_jitterbuffer_msec=100"/>
      <action application="set" data="RECORD_STEREO=true"/>
      <action application="record_session" data="/tmp/mytestingfile.wav"/>
      <action application="bind_meta_app" data="2 ab s lua::prtask_pause.lua"/>
      <action application="bind_meta_app" data="3 ab s lua::prtask_resume.lua"/>
      <action application="bridge" data="sofia/external/${destination_number}@$$
    </condition>
  </extension>

Save this is add lua scripts at /usr/share/freeswitch/scripts/ .

Make a script prtask_pause.lua and add the following script to it

CODE

local filename = "/tmp/mytestingfile.wav";
freeswitch.consoleLog("INFO", "==============================================Pa$
session:execute("record_session_pause", filename)

Save this and make another script prtask_reume.lua and add the following script to it

CODE

local filename = "/tmp/mytestingfile.wav";
freeswitch.consoleLog("INFO", "==============================================Pa$
session:execute("record_session_resume", filename)

Save this. Now run reloadxml in fs_cli. Pause the call recording by pressing *2 and resume call recording by pressing *3 from any end side.