---
title: "Ring2All SBC - RTPEngine Media Relay & WebRTC Gateway Documentation"
description: "Documentation for RTPEngine Media Relay"
---

## Table of Contents
1. [Module Overview (Technical)](#1-module-overview-technical)
2. [Module Overview (Commercial/Business)](#2-module-overview-commercialbusiness)
3. [Module Overview (End User/Administrator)](#3-module-overview-end-useradministrator)
4. [Configuration Sections](#4-configuration-sections)
5. [Settings Reference](#5-settings-reference)
6. [Common Scenarios & Examples](#6-common-scenarios--examples)
7. [Limitations & Important Notes](#7-limitations--important-notes)
8. [Troubleshooting Tips](#8-troubleshooting-tips)
9. [Glossary](#9-glossary)

---

## 1. Module Overview (Technical)

### What Is the RTPEngine Media Relay Module?

The **RTPEngine Media Relay & WebRTC Gateway Module** in Ring2All SBC manages real-time audio/video proxying, NAT traversal, SRTP encryption/decryption, and WebRTC-to-SIP bridging (ICE, DTLS-SRTP, Opus/G.711 codec transcoding).

Using kernel-space packet forwarding (`xt_RTPENGINE`), Ring2All SBC routes media streams directly through the Linux kernel network stack, achieving high-throughput, sub-millisecond audio latency for thousands of concurrent channels.

### Architecture

```
┌─────────────────────────────────────────────────────────────────┐
│            Ring2All SBC - RTPEngine Architecture                │
├─────────────────────────────────────────────────────────────────┤
│                                                                 │
│  WebRTC Client (Browser)          PSTN / Telephony Server             │
│  WSS / ICE / DTLS-SRTP            UDP / RTP (G.711 / G.729)    │
│            │                                  │                 │
│            ▼                                  ▼                 │
│  ┌──────────────────────────────────────────────────────────┐   │
│  │                    Ring2All SBC                           │   │
│  │  ┌──────────────────┐          ┌──────────────────────┐  │   │
│  │  │ Kamailio Signaling│ <----->  │ RTPEngine Media Proxy│  │   │
│  │  └──────────────────┘          └──────────────────────┘  │   │
│  │                                  (xt_RTPENGINE Kernel)   │   │
│  └──────────────────────────────────────────────────────────┘   │
│                                                                 │
└─────────────────────────────────────────────────────────────────┘
```

---

## 2. Module Overview (Commercial/Business)

### Business Benefits

1. **Seamless WebRTC Integration**: Enables softphones, browsers, and mobile apps to communicate directly with traditional SIP trunks without expensive hardware SBCs.
2. **Sub-Millisecond Latency**: Kernel-level media forwarding guarantees crystal-clear voice quality without CPU bottlenecking.
3. **Enterprise Encryption & Security**: Enforces mandatory DTLS-SRTP encryption for remote endpoints over public Internet.

---

## 3. Module Overview (End User/Administrator)

### What Can You Do in This Module?

- **Configure Media Proxies**: Set listening interfaces, UDP port ranges (`10000-20000`), and TOS/QoS audio flags.
- **Enable WebRTC Gateway**: Auto-bridge WebSockets (ICE/SDP) to traditional UDP SIP.
- **Monitor Active Media Sessions**: Track active call streams, jitter, packet loss, and codec stats.

---

## 4. Configuration Sections

### 1. RTPEngine Service Settings
Configures daemon socket control, kernel module bindings, and thread pools.

### 2. Media Bridge Rules
Defines rules for converting SRTP to RTP, rewriting SDP IPs, and managing ICE candidates.

---

## 5. Settings Reference

| Field Name | Type | Options / Format | Description |
| :--- | :--- | :--- | :--- |
| `listen_ng` | Input | `IP:Port` (e.g. `127.0.0.1:2223`) | Control socket for Kamailio-to-RTPEngine communication. |
| `port_min` | Integer | `10000` | Minimum UDP port for media streams. |
| `port_max` | Integer | `20000` | Maximum UDP port for media streams. |
| `rtpengine_flags` | String | `trust-address replace-origin replace-session-connection` | SDP manipulation parameters. |

---

## 6. Common Scenarios & Examples

### Scenario: WebRTC Browser to PSTN Call Bridging

1. WebRTC client connects via WebSocket (`wss://sbc.ring2all.xyz`).
2. Kamailio invokes `rtpengine_offer("ICE=force DTLS=passive SDES-off RTP/SAVPF")`.
3. RTPEngine negotiates DTLS-SRTP with browser and plain RTP with PSTN gateway.
4. Media flows smoothly with zero audio degradation.

---

## 7. Limitations & Important Notes

- Ensure `xt_RTPENGINE` kernel module is loaded for maximum performance (`modprobe xt_RTPENGINE`).

---

## 8. Troubleshooting Tips

| Symptom | Probable Cause | Solution |
| :--- | :--- | :--- |
| One-way audio | NAT / Public IP mismatch in SDP | Verify `advertised_ip` in RTPEngine config. |
| WebRTC call drops instantly | ICE candidate negotiation failure | Ensure TURN/STUN servers or public ICE IPs are configured. |

---

## 9. Glossary

- **ICE**: Interactive Connectivity Establishment for WebRTC NAT traversal.
- **DTLS-SRTP**: Datagram Transport Layer Security for SRTP media key exchange.