bitspath-sip~3,500 locSIP parser, transactions, transports (UDP/TCP/TLS/WebSocket). RFC 3261 compliant message parsing with zero-copy headers.
Voice PBX: pure Rust SIP stack
An enterprise PBX, rebuilt from the SIP parser up.
A complete PBX built from scratch in Rust: SIP parser, B2BUA, RTP relay, conference mixer. 20,000 lines replacing 2 million. No FreeSWITCH, no Asterisk.
- Memory-safe by construction
- Native multi-tenant (RLS)
- REST API for everything
console.bitspath.com/voice
ext 204+8801•• support linerec04:12Talking
ext 118conference: standup12:47Mixing
trunk-02IVR: main menu00:19DTMF
Queue: support2 agents ready
12
Active calls
0:42
Avg wait
On
SRTP
20K
Lines of Rust
4
Purpose-built crates
255+
REST endpoints
35+
RLS tables
12 core modules
Everything a modern PBX needs
All managed through the Portal UI and the REST API. No config files to hand-edit, no dialplan scripting language to learn.
01
Extensions
CRUD, bulk import/export, provisioning profiles for SIP phones and softphones.
02
Ring Groups
4 hunt strategies: simultaneous, sequential, round-robin, weighted routing.
03
IVR Menus
Multi-level menus, DTMF routing, time-of-day conditions, failover destinations.
04
Call Queues
Agent management, configurable hold music, max wait time, queue callbacks.
05
Voicemail
Custom greetings, message management, email notifications with audio attachments.
06
Conferences
PIN-protected rooms, moderator controls, mute/kick, participant limits.
07
Call Flows
Visual drag-and-drop flow builder with conditions, branches, and sub-flows.
08
Routes
Inbound/outbound routing rules, time conditions, regex pattern matching, trunks.
09
CDR
Call detail records with advanced filters, date ranges, and CSV export.
10
Recordings
On-demand and automatic call recording, streaming playback, bulk download.
11
Music on Hold
Multiple categories, track management, per-queue and per-tenant playlists.
12
Active Calls
Real-time call monitoring dashboard: listen, whisper, barge, hangup, transfer.
Architecture, not marketing
How BitsPath Voice compares
Side-by-side with the incumbents on the things that decide operability: codebase, tenancy, API surface, and memory safety.
| Feature | BitsPath Voice | FreeSWITCH | Asterisk | 3CX |
|---|---|---|---|---|
| Language | Rust | C | C | C++ (proprietary) |
| Codebase Size | ~20,000 lines | 2,000,000+ lines | 1,000,000+ lines | Proprietary |
| License | Platform subscription | MPL 1.1 / Commercial | GPL v2 / Commercial | Per-user license |
| SIP Stack | bitspath-sip (custom) | mod_sofia (oSIP) | chan_pjsip (PJSIP) | Proprietary |
| Media Engine | bitspath-media (SRTP, G.711/G.722) | mod_dptools | Built-in | Built-in |
| Web Admin UI | Included (Portal) | FusionPBX (separate) | FreePBX (separate) | Included |
| Multi-Tenant | Native RLS (35+ tables) | Domain-based (add-on) | Not native | Limited |
| REST API | 255+ endpoints | ESL + mod_xml_curl | AMI / ARI | Proprietary API |
| WebRTC | WebSocket transport | mod_verto | Built-in (limited) | Built-in |
| Memory Safety | Guaranteed (Rust) | Manual (C) | Manual (C) | Unknown |
Four purpose-built crates
No forks, no wrappers, no FFI bindings
Every line is ours, which is why the whole stack fits in four crates you can actually audit.
bitspath-pbx~6,650 locRegistrar, routing engine, CDR, call forwarding, ring groups, IVR menus, voicemail, and conference bridge orchestration.
bitspath-media~3,000 locRTP/RTCP/SRTP processing, G.711 and G.722 codecs, adaptive jitter buffer, conference mixing with N-1 audio streams.
bitspath-call~7,000 locB2BUA call engine, call state machine, RTP relay with NAT latching, reinvite handling, and oRTP-based media relay.
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