SipRecepta

Incoming SIP call router for SIP-Jibri.

This repository contains scripts, files and documentation to convert FreeSwitch into a router for incoming SIP calls for SIP-Jibri. Assumed that following services are already installed and configured correctly:

• Jitsi
• Jitsi-Component-Selector
• SIP-Jibri with Jitsi-Component-Sidecar
• FreeSwitch

Adding SipRecepta

SipRecepta is a plugin for FreeSwitch to route incoming SIP calls to SIP-Jibri. The following steps are tested in Debian 12 Bookworm. Assumed that FreeSwitch already installed and running on this server.

Install additional packages

apt-get update
apt-get install python3 python3-requests

Enable Python3 in FreeSwitch

In /etc/freeswitch/autoload_configs/modules.conf.xml:

<load module="mod_python3"/>

Install SipRecepta

cd /usr/local/lib/python3.11/dist-packages
mkdir nordeck
cd nordeck
touch __init__.py
wget https://raw.githubusercontent.com/nordeck/sip-recepta/main/files/siprecepta.py

DialPlan

cd /etc/freeswitch/dialplan/public
wget https://raw.githubusercontent.com/nordeck/sip-recepta/main/files/98_public_siprecepta_dialplan.xml

cd /etc/freeswitch/dialplan/default
wget https://raw.githubusercontent.com/nordeck/sip-recepta/main/files/99_default_siprecepta_dialplan.xml

Directory

Put siprecepta folder into /etc/freeswitch/directory/default.xml

<groups>
  <group name="default">
    <users>
      <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
      <X-PRE-PROCESS cmd="include" data="/tmp/siprecepta/*.xml"/>
    </users>
  </group>
</groups>

Global variables

Put the following variables into /etc/freeswitch/vars.xml:

<X-PRE-PROCESS cmd="set" data="component_selector_url=https://domain/path"/>
<X-PRE-PROCESS cmd="set" data="component_selector_verify=false"/>
<X-PRE-PROCESS cmd="set" data="component_selector_token=eyJhbG..."/>

component_selector_verify should be false if Jitsi-Component-Selector has not a trusted certificate.

component_selector_token is needed if PROTECTED_SIGNAL_API is enabled for Jitsi-Component-Selector. Generate the token by running the following commands:

PRIVATE_KEY_FILE="/path/to/signal.key"

HEADER=$(echo -n '{"alg":"RS256","typ":"JWT","kid":"jitsi/signal"}' | \
  base64 | tr '+/' '-_' | tr -d '=\n')
PAYLOAD=$(echo -n '{"iss":"signal","aud":"jitsi-component-selector"}' | \
  base64 | tr '+/' '-_' | tr -d '=\n')
SIGN=$(echo -n "$HEADER.$PAYLOAD" | \
  openssl dgst -sha256 -binary -sign $PRIVATE_KEY_FILE | \
  openssl enc -base64 | tr '+/' '-_' | tr -d '=\n')

TOKEN="$HEADER.$PAYLOAD.$SIGN"
echo $TOKEN

Restart

Restart the service:

systemctl restart freeswitch

Usage

Call the number set in 98_public_siprecepta_dialplan.xml. e.g.:

112233@freeswitch_address:5080

Type PIN when asked. For now there is a hardcoded value which is 123456.

Tips

Default port

If it is possible for this setup to set 5060 as the external SIP port then callers can use SIP address without setting the port.

This is fine if you have only an IP address for this setup. If you have a domain name, use the following...

DNS records

If you have a domain name, set SRV records for SIP service.

• DNS A record for sip.mydomain.com which points the IP address of this server.

• DNS SRV record for _sip._tcp.mydomain.com which points sip.mydomain.com. Select 5080 as port if you still use the default external port.

• DNS SRV record for _sip._udp.mydomain.com which points sip.mydomain.com. Select 5080 as port if you still use the default external port.

In this case, the remote peer can call 112233@mydomain.com without setting the port.

You may check the records in VoIP Toolbox. Write your domain into the input box. e.g mydomain.com