Run sipnab beside OpenSIPS

sipnab reads the SIP that passes through OpenSIPS off the wire, so it needs nothing from OpenSIPS: no module, no configuration change. On the OpenSIPS machine it sees both legs of every call, the caller’s side into OpenSIPS and OpenSIPS’s side out to the callee, and puts them together into one call.

This guide adds sipnab to the OpenSIPS proxy from Use OpenSIPS as your voice stack’s SIP server, installed from the packages or built from source.

Tested on

Every command on this page ran as written, in order, on 2026-09-27, with sipnab 0.5.193 from its release package, on the machines the OpenSIPS guide had set up: Debian 13 (kernel 6.12.63) with OpenSIPS built from source, and Ubuntu 24.04.5 (kernel 6.8.0) with OpenSIPS both from the packages and from source. The section with Kamailio on the same machine ran on Ubuntu 24.04.5. The examples use 192.0.2.10 as the machine’s address.

1. Install sipnab

# Run all of these, in order.
V=$(curl -fsSL https://api.github.com/repos/NormB/sipnab/releases/latest \
  | python3 -c 'import json,sys; print(json.load(sys.stdin)["tag_name"].lstrip("v"))')
curl -fsSLO https://github.com/NormB/sipnab/releases/download/v$V/sipnab_${V}_amd64.deb
curl -fsSL https://github.com/NormB/sipnab/releases/download/v$V/SHA256SUMS.txt \
  | grep " sipnab_${V}_amd64.deb$" | sha256sum -c -
sudo apt-get install -y ./sipnab_${V}_amd64.deb
sipnab --version

2. Watch a call through OpenSIPS

Start sipnab on every interface for 30 seconds, printing each SIP message as it passes and a report of the calls at the end:

sudo sipnab -N -d any --duration 30 --report

While it runs, place a test call from a second terminal, as in step 3 of the OpenSIPS guide:

# Run all of these, in order.
cd ~/sipp
sipp -sn uas -i 127.0.0.1 -p 5070 -m 1 -bg
sipp -sf uac_rr.xml 192.0.2.10:5060 -i 192.0.2.10 -p 5080 -m 1 -d 1000 -timeout 20s

sipnab prints every message OpenSIPS relays twice, once on each leg: the caller’s INVITE to OpenSIPS at 192.0.2.10:5060, then OpenSIPS’s copy of it to the callee at 127.0.0.1:5070, and the same for the replies and the BYE. OpenSIPS’s own 100 Trying goes to the caller alone. When the 30 seconds are up, the report lists the call once, as Completed, with 13 messages from both legs.

For the same view in a terminal interface, which updates as calls happen, run sudo sipnab -d any without -N and without --duration.

With Kamailio on the same machine

With OpenSIPS on 5060 and Kamailio on 5062, as OpenSIPS and Kamailio on one machine sets them up, give sipnab both ports:

sudo sipnab -N -d any --duration 30 --report --portrange 5060-5062

A call through each proxy then appears in the report, one row per call.

When something does not work

• sipnab reports No SIP traffic found. OpenSIPS listens on a port outside 5060-5061, the range sipnab watches by default. Give sipnab OpenSIPS’s ports with --portrange.