SIP Routing & SBC

OpenSIPS — the routing brain of your voice network

OpenSIPS sits in front of every call: it load-balances media nodes, routes to the right carrier, handles WebRTC registrations and keeps unwanted SIP traffic out. We design and operate it for contact centers and carriers.

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  • Load balancing
  • SBC with rtpengine
  • Caller-ID / prefix routing
  • WebRTC registrar

Media servers shouldn't face the internet

Exposing Asterisk or FreeSWITCH directly to carriers and agents makes every box a security and routing problem. Scaling means editing trunks on each server.

With OpenSIPS in front, there is one place for routing policy, security and registration — and media servers become interchangeable.

How It Works

What OpenSIPS does on every call

01

Authenticate & filter

Only allow-listed carriers and authenticated users get through; scanners are dropped.

02

Route

Match caller-ID ranges, prefixes and trunk capabilities to pick a destination.

03

Balance & anchor

Select a healthy media node and anchor RTP through rtpengine where needed.

Features

OpenSIPS features we implement

Load balancing

Distribute calls across Asterisk/FreeSWITCH nodes with health probing.

Dynamic routing

Database-driven routes by caller-ID range, prefix, campaign or tenant.

SBC & topology hiding

Hide internal topology, normalise headers and control PAI/RPID per carrier.

rtpengine media relay

NAT traversal, multi-interface media and optional recording hooks.

WebRTC registrar

WSS registrations for browser agents, with stale-contact cleanup.

Fraud protection

Rate limits, IP allow-lists and destination restrictions.

Architecture

OpenSIPS in the stack

One routing tier for carriers, agents and media nodes.

  1. Carriers & agents
    SIP / WSS
  2. OpenSIPS
    Auth, route, LB
  3. rtpengine
    Media relay
  4. Media nodes
    Asterisk / FreeSWITCH

Compliance & Security

Security at the edge

IP allow-listing for carrier trunks
Rate limiting and anti-scanner rules
Topology hiding for internal servers
TLS/WSS for browser signalling
Per-tenant and per-trunk concurrency caps
Centralised SIP logging for troubleshooting

FAQ

Common questions

Do we need OpenSIPS if we have one Asterisk server?+
Not always — but as soon as you have multiple carriers, WebRTC agents or more than one media node, an SBC/routing tier pays for itself.
OpenSIPS or Kamailio?+
Both are capable SIP proxies. We specialise in OpenSIPS and its modules for load balancing, dynamic routing and WebRTC.
Can OpenSIPS route by caller-ID?+
Yes. Routes can match the presented number range and send the call to the trunk or gateway that owns that range.
Does it handle WebRTC?+
Yes. OpenSIPS acts as the WSS registrar for browser softphones, with rtpengine bridging WebRTC and SIP media.

Related

C2C Connectivity WebRTC Softphone ITSP / Carrier

Get Started

Put a proper SBC in front of your voice network

We'll review your trunks and nodes and propose an OpenSIPS design.

Talk to an OpenSIPS engineer Get a quote →
+91 98991 52832WhatsApp