Communications·production

IAM.Comm

Carrier-grade media and signalling stack for SIP, IMS, WebRTC, RTP and recording.

Landing Application

Key capabilities

  • SIP/IMS signalling and media gateway
  • WebRTC, conferencing and recording
  • ASR, TTS and VAD plugins with a post-call pipeline

Step-by-step guides

Use cases

Start with the outcome: open a guide, prepare prerequisites, follow the steps and verify the success signals.

01 Connect a SIP trunk and verify an inbound callThe call traverses signalling and media and is recorded under one call id.
Audience
Telecom engineer
Outcome
The call traverses signalling and media and is recorded under one call id.

Before you start

  • SIP peer parameters
  • Allowed IPs/ports
  • Test DID and codec list

Steps

  1. Create the trunkSet peer, auth mode, transport, codecs and number normalization.
  2. Configure networkAllow signalling and RTP ranges; verify NAT/public address.
  3. Bind a routeLink the DID to an inbound scenario, recording policy and Voice/Speech pipeline.
  4. Place a test callVerify 200 OK/ACK, two-way RTP and clean BYE termination.
Verify the result

Call detail record, media metrics and recording share one call id.

If it does not work

One-way audio usually indicates NAT/RTP allowlist issues rather than a Voice scenario.

02 Enable recording and post-call processingRecording follows policy and reaches Speech with consent metadata intact.
Audience
Platform operator
Outcome
Recording follows policy and reaches Speech with consent metadata intact.

Before you start

  • Retention/consent policy
  • Object storage
  • IAM.Speech endpoint

Steps

  1. Create a recording policySet channels, consent requirement, encryption and retention.
  2. Attach to the routeApply the policy only to required queues/DIDs and exclude test numbers.
  3. Configure post-call hookSend a signed object reference, call id, tenant and consent status.
  4. Verify lifecycleConfirm recording ready, Speech accepted and retention deletion schedule.
Verify the result

Recording is visible only to permitted roles and Speech job links to the call id.

If it does not work

Do not send recording binaries through the webhook; use a scoped signed reference.

Application sections

Open detailed manual

Every application screen has a separate page with controls, safe example values, CLI/API alternatives and status-specific recovery steps.

Open detailed manual →

Role in the ecosystem

IAM.Comm - operator communication layer. It connects signaling, media and AI plugins into an observable circuit that can be deployed on-premise, dedicated or in a protected Kubernetes perimeter.

Platform composition

  • SIP/IMS ingress, registration and routing of sessions;
  • RTP/WebRTC media gateway and codec negotiation management;
  • recording, conference/mixing and media events;
  • plugins VAD, ASR, TTS and post-call processing;
  • console for the status of sessions, nodes and quality.

Call flow

Signalling creates a session and negotiates media parameters. Media plane transmits streams without outputting sensitive audio to the control plane. Connected AI modules receive only allowed segments, and events and correlation identifiers allow you to restore the call path.

Safety and Operation

The perimeter defines the available SIP trunks, codec, entry and retention. For diagnostics monitor packet loss, jitter, RTT, MOS approximation, registration and reasons for termination. Recording and transcription are enabled by policy, taking into account local requirements and consent of participants.

IAM.Comm Meet uses WebRTC conferences, IAM.Voice - voice scripts, and IAM.Speech - analysis of recordings and conversations.