I build and debug the parts of telephony that carry actual calls: SIP proxies, media relays, and the tooling around them. I have been contributing upstream since 2006. Most of my work is on OpenSIPS, Kamailio, rtpengine, FreeSWITCH and Asterisk — upstream where it belongs, and in my own tools where nothing suitable existed.

I also run this stack in production, which is where most of the bugs come from: OpenSER since 2006, then OpenSIPS and Kamailio, and FreeSWITCH since 2008.

20 years contributing upstream (since 2006) · 75 merged pull requests across 10 upstream projects · ~20 upstream bug reports, 4 of them crashes · 75,000+ installs of my two language servers · 12 conference talks, 2007–2026

Project	What it is
sipnab · sipnab.com	SIP & RTP capture, analysis and security. One binary, one dependency (libpcap).
vcon-server · vcon-server.com	Stores IETF vCons encrypted at rest — AES-256-GCM, a key per version wrapped by tenant and master keys — with tenants, retention, legal holds and append-only audit logs. 0.4.1 released, not yet for production.
opensips-lsp · Open VSX	Language server for OpenSIPS routing scripts — completion, diagnostics, go-to-definition, rename across every use, outline and folding. Its tree-sitter grammar parses every config that opensips -C 4.0.2 accepts.
kamailio-lsp · Open VSX	The same, for Kamailio configuration files. Follows the #!ifdef preprocessor as Kamailio does; its grammar parses every shipped Kamailio 6.1.5 config that kamailio -c accepts.
vcon-datasets-core04	Three public vcon-dev vCon datasets (16,799 containers) corrected to draft-ietf-vcon-vcon-core-04 by additive, rule-checked edits, reproducibly, with the open items listed. Before: 0 passed the core-04 schema; after: all 8,503 Supreme Court and 4,527 of 8,181 IETF containers. Fixes offered upstream as three open pull requests (Supreme Court · IETF · Newport).
devstack-core	Local VoIP development infrastructure — Docker, Postgres, MongoDB, RabbitMQ, Prometheus, Grafana.
voip-stack	Production-shaped OpenSIPS + Asterisk + rtpengine deployment.
opensips-experiment	An OpenSIPS module written in Rust — the experiment behind the 2023 Summit talk.
homebrew-tap	brew install NormB/tap/sipnab
mod_mosquitto	FreeSWITCH ↔ MQTT event handler. Lives upstream under the FreeSWITCH org.

Open upstream. NATS support for OpenSIPS is an open pull request, OpenSIPS/opensips#4442: five community-contributed modules under modules-contrib/ — nats, event_nats, cachedb_nats, cachedb_nats_fts and nats_consumer — with development continuing on feature/nats (672 commits ahead of upstream).

Not yet upstream. One branch lives in my OpenSIPS fork while it matures:

Two decades of features and fixes that landed in other people's projects, oldest first.

By year: 9 merged in 2024 · 13 in 2025 · 37 in 2026 — the stretch covered by the 2026 Summit talk on AI-assisted development. The method is not delegation: I hold the design and the judgment, and use AI for the heavy lifting — correlating days of logs across dozens of servers into diagnosable patterns, and building the analysis scripts that follow.

Counts hide the interesting part. The OpenSIPS work alone spans at least a dozen modules — cachedb_redis, dialog, tm, janus, rtpengine, b2b_entities, pua_dialoginfo, stir_shaken, sockets_mgm, dispatcher, event_rabbitmq, cfgutils — and is mostly memory-safety and concurrency work. A sample:

• Redis Cluster support for OpenSIPS — hash-tag support (#3815), safer redirect parsing (#3854), dynamic topology management (#3855) and Unix-socket transport with lazy connect (#3856)
• WebSocket fragment reassembly in the Janus module — RFC 6455 §5.4; fragmented frames were previously rejected outright
• Use-after-free and race in dialog cluster replication, and a race in sockets_mgm receive_fd that spun the process on reload
• STIR/SHAKEN: strip RFC 4904 user parameters from telephone numbers
• FreeSWITCH core — switch_rtp max-missed-packets handling, XML generation, a mod_v8 compile warning and switch_core parameter docs
• Password handling in valkey-admin — decrypt passthrough and double-encryption fixes
• SIP→WebRTC transcoding regression tests — four sipssert scenarios pinning down rtp_relay issue #3902, contributed to the project's own test suite

Roughly twenty defects found while running this stack in production and reported upstream, with reproductions. Four were crashes:

• sipnab — deepening protocol coverage (SIP, SDP, RTP/RTCP, STUN/TURN, TLS/DTLS) and the MCP server that exposes it to agents, now also served over HTTP beside the TUI; vCon output to draft-ietf-vcon-vcon-core-04, with --vcon-fetch to read stored vCons back; step-by-step guides for running it beside OpenSIPS, Kamailio, Asterisk, FreeSWITCH, rtpengine, rtpproxy, Homer, Prometheus, fail2ban, a vCon server and TFPS, every step run on Debian 13 and Ubuntu 24.04; and a code-quality program that removes every lint suppression by redesigning the code it hid, and converts the tests to return their errors rather than panic
• vcon-server — the store, the HTTP API, tenants and their keys, retention and the purge sweep, legal holds, the audit logs, the command line, key rotation and check are built and tested, with S3 and Parquet storage; 0.4.1 is released, not yet for production; an optional PostgreSQL catalog is in progress
• opensips-lsp / kamailio-lsp — bringing real editor tooling to SIP routing scripts, which have never had any; the October and November releases (now 0.28.1 / 0.22.1) make scanning linear in document size, keep answering requests while a config check runs, add outline, folding and selection by structure, and read $(...) variables the way the real parsers do, proven against OpenSIPS 4.0.2 and Kamailio 6.1.5; kamailio-lsp resolves includes in the order Kamailio opens them
• vCon — testing sipnab's vCon schema validator against the public vcon-dev datasets, and correcting those datasets to core-04 (vcon-datasets-core04), with the corrections offered back upstream, fake-vcons included (#3); and an open issue on the core draft, where content_hash cannot be computed for a recording referenced by a streaming-service page
• OpenSIPS NATS modules — five modules-contrib modules, open upstream as #4442
• OpenSIPS upstream — reproducing and fixing memory-safety and concurrency bugs (use-after-free, races, leaks) across the dialog, tm, Redis and media modules, and writing regression tests so they stay fixed — including in OpenSIPS' own SIPssert test framework
• devstack-core — making a full VoIP stack reproducible on an ARM laptop

Speaking on SIP and real-time media since 2007, mostly at the OpenSIPS Summit and ClueCon.

Technical reviewer for FreeSWITCH 1.2 (Packt Publishing, second edition, May 2013, ISBN 978-1-78216-100-4).

Member of the OpenSIPS Foundation, listed on its member page (last updated December 2014).

SIP stacks     OpenSIPS · Kamailio · Asterisk · FreeSWITCH
Media          rtpengine · RTP/RTCP · SRTP · SDP · codecs
Languages      Rust · C · Shell · Python
Data & bus     Valkey/Redis · PostgreSQL · MongoDB · RabbitMQ · NATS
Ops            Docker · Prometheus · Grafana · HAProxy · libpcap
Testing        SIPp · tcpdump/pcap analysis · fault reproduction

sipnab.com · [email protected] · linkedin.com/in/nbrandinger

For anything about a specific project, an issue on that repository reaches me fastest.