Session Router signaling and media plane

SIP server, media engine, rule engine, and CPaaS runtime

Programmable voice infrastructure for real SIP networks.

Session Router combines carrier-facing SIP routing, B2BUA dialog control, native RTP media handling, and webhook-driven CPaaS verbs in one server. It is built for teams that need call control close to the signaling path, not a detached application layer.

Why Session Router

One control point for routing, media, and programmable voice.

Session Router is aimed at service providers, communications platforms, and voice teams that need to operate SIP traffic and expose higher-level call control without splitting responsibility across several products.

Reduce signaling complexity

Route calls by tenant, account, profile, trunk, user, headers, and dialed identity from one rule model.

Keep media under control

Anchor, bridge, play, synthesize, record, and stream audio where call behavior requires it.

Expose programmable voice

Let applications drive live calls using HTTP webhooks and JSON CPaaS verbs.

Core engines

The platform is more than a SIP listener.

Session Router is organized around engines that each own a specific part of the voice path: routing decisions, SIP dialog state, media, CPaaS call control, queueing, and clustered operations.

RULE ENGINE

Routing decisions and dial-plan policy

Evaluates routing profiles and ordered rules to decide whether a call is allowed, rejected, routed to a local user, sent to a trunk, sent to a trunk group, proxied, or handed to a CPaaS application. It also supports sequential and parallel routing, crankback/reroute behavior, and ordered header manipulation with Store, Add, Modify, and Delete actions.

SIP B2BUA

Dialog, transaction, and bridge control

Handles SIP over UDP, TCP, TLS, WS, and WSS, with transaction matching, digest authentication, per-dialog worker serialization, linked-leg forwarding, glare handling, PRACK/100rel support, and call teardown.

MEDIA ENGINE

RTP, SRTP, playback, and streaming

Manages RTP/RTCP, DTLS-SRTP, SDES-SRTP, ICE/STUN for WebRTC legs, MRF playback, TTS audio injection, DTMF, recording paths, and bidirectional WebSocket media streaming.

CPAAS RUNTIME

Webhook-controlled call flows

Processes JSON verb arrays returned by application webhooks to answer, speak, play, gather input, dial, bridge, enqueue, redirect, stream, send DTMF, and send in-dialog SIP requests.

QUEUE ENGINE

Call queueing and agent hunt

Places calls into named queues, tracks position and elapsed queue time, drives wait treatment through restricted verbs, and can hunt for the next available agent.

CONTROL PLANE

Admin API, configuration, and clustering

Provides a Postgres-backed admin API and UI, CDR storage, settings, node/controller cluster endpoints, configuration snapshots, and delta propagation for multi-node deployments.

SIP Server features

Built for operational SIP routing, not only call origination.

The SIP server layer handles the carrier and endpoint realities that application platforms often hide: registration, authentication, routing policy, trunk failover, media negotiation, and safe per-call concurrency.

Multi-transport SIP

UDP, TCP, TLS, WS, and WSS listener support for trunks, endpoints, and browser-oriented SIP stacks.

Registration control

Digest authentication, registration rules, endpoint NAT handling, and registered-user routing.

Routing profiles

Hierarchical rule sets can be scoped across tenant, company, account, profile, user, and trunk layers.

Trunks and trunk groups

Route to carriers and grouped trunks with timeouts, failover behavior, crankback controls, and media options.

Header manipulation

Match and transform SIP headers or request URI fields using ordered Add, Modify, Delete, and Store operations.

Media security

Support for RTP, SRTP modes, DTLS, ICE/STUN behavior, codec negotiation, and WebRTC-oriented media paths.

CPaaS features

Application-controlled calls without leaving the SIP server.

CPaaS applications receive call events and return JSON instructions. Session Router then executes the verbs on the live SIP session, keeping application logic simple while the server owns the call state and media path.

CALL EVENT Session Router posts call state to the configured application hook.
JSON VERBS The application responds with an ordered array of call-control verbs.
SERVER EXECUTION The verb engine answers, plays, gathers, dials, bridges, queues, or redirects.
NEXT ACTION Action hooks can return another verb array to continue the call flow.

Documentation split

Website for value. Docs for implementation.

The marketing website stays descriptive and benefit-led. The docs are separate static pages with low-level developer details, runtime behavior, examples, and implementation notes.