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Ring2All Documentation

Ring2All SBC - RTPEngine Media Relay & WebRTC Gateway Documentation

3 min readUpdated: Sep 26, 2026
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Table of Contents

Section titled “Table of Contents”
  1. Module Overview (Technical)
  2. Module Overview (Commercial/Business)
  3. Module Overview (End User/Administrator)
  4. Configuration Sections
  5. Settings Reference
  6. Common Scenarios & Examples
  7. Limitations & Important Notes
  8. Troubleshooting Tips
  9. Glossary

1. Module Overview (Technical)

Section titled “1. Module Overview (Technical)”

What Is the RTPEngine Media Relay Module?

Section titled “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

Section titled “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)

Section titled “2. Module Overview (Commercial/Business)”

Business Benefits

Section titled “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)

Section titled “3. Module Overview (End User/Administrator)”

What Can You Do in This Module?

Section titled “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

Section titled “4. Configuration Sections”

1. RTPEngine Service Settings

Section titled “1. RTPEngine Service Settings”

Configures daemon socket control, kernel module bindings, and thread pools.

2. Media Bridge Rules

Section titled “2. Media Bridge Rules”

Defines rules for converting SRTP to RTP, rewriting SDP IPs, and managing ICE candidates.

5. Settings Reference

Section titled “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

Section titled “6. Common Scenarios & Examples”

Scenario: WebRTC Browser to PSTN Call Bridging

Section titled “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

Section titled “7. Limitations & Important Notes”
  • Ensure xt_RTPENGINE kernel module is loaded for maximum performance (modprobe xt_RTPENGINE).

8. Troubleshooting Tips

Section titled “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

Section titled “9. Glossary”
  • ICE: Interactive Connectivity Establishment for WebRTC NAT traversal.
  • DTLS-SRTP: Datagram Transport Layer Security for SRTP media key exchange.