RingQ Customer Centric Communications

RingQ Network Codec Configuration


RingQ PBX Platform

Network & Codec Configuration Guide

General Settings

Contact Center

Network

This guide covers the two core configuration areas inside the RingQ Network settings dialog: IP configuration (network interface selection and external IP/NAT setup) and Codec configuration (negotiation strategy, transcoding, and codec priority). Correct settings in both areas ensure clear, reliable calls across all SIP trunk providers and connected endpoints.

Open General Settings and locate the Contact Center section. Click the Network tile (highlighted below) to open the Network configuration dialog.

General Settings — Network tile highlighted in the Contact Center section

The Network dialog contains two tabs: IP and Codec. Each tab controls a distinct aspect of the PBX’s communication stack.

1. IP Tab — Network Interface & External IP

The IP tab determines which server network card RingQ listens on for SIP and RTP traffic, and what public IP address it advertises to remote SIP endpoints for correct media routing.

Network dialog — IP tab showing NIC dropdown and External IP field

1.1 Select Network Card Interface

This dropdown lists all network interface cards (NICs) detected on the server. Each entry shows the interface name and its assigned IP, for example 165.232.169.74 (eth0). The selected interface is where RingQ binds its SIP listener and RTP media engine.

What this setting controls

RingQ binds its SIP listener and RTP media engine to the IP of the selected interface.

All SIP REGISTER, INVITE, and RTP media packets flow through this interface.

This must be the Default Internet-facing interface (Default gateway interface).

When to change the NIC selection

  • Your server has multiple NICs — choose the one that faces your SIP trunk provider’s network.
  • You want to isolate VoIP traffic from management traffic on a dedicated interface.
  • After adding a NIC or reassigning IPs — select the interface carrying your public or routable IP.

⚠ Warning

Selecting an internal-only NIC without setting the External IP field will cause one-way audio or SIP registration failures on external calls.

1.2 External IP Configuration

Enter the public-facing IP address of your server or NAT gateway. RingQ embeds this value into outbound SIP headers (Contact, Via) and SDP connection lines so that remote endpoints know where to send return audio.

Scenario

What to enter

Server behind a home / office router

The WAN (public) IP of your router — verify via whatismyip.com from the server

Cloud VM (AWS, DigitalOcean, etc.)

The Elastic IP or floating IP assigned to the instance, not the private 10.x or 172.x address

Server with a direct public IP on NIC

May match the NIC IP; confirm with: curl ifconfig.me

Multiple SIP trunks from different providers

A single External IP is sufficient if all trunks share the same public route

⚠ Dynamic IPs

If your public IP changes (dynamic ISP), update this field whenever the IP changes, or use a DDNS hostname maintained by your firewall or router.

2. Codec Tab — Negotiation, Transcoding & Priority

The Codec tab controls how RingQ negotiates audio encoding with SIP trunks and endpoints, and whether it converts audio between codec formats in real time during a call.

Network dialog — Codec tab with Codec Negotiation dropdown expanded

2.1 Codec Negotiation

This dropdown controls the strategy RingQ uses when offering or accepting codecs during SIP call setup (SDP offer/answer). Four modes are available:

Mode

Behaviour and best use

Adaptive

Dynamically selects the best mutually-supported codec from both sides. Recommended for most deployments — flexible and fault-tolerant.

Strict

Only accepts codecs from RingQ’s priority list. Calls fail if no match is found. Use for compliance, recording, or quality-controlled environments.

Low-bandwidth

Prefers narrowband / low-bitrate codecs (G.729, PCMA) to conserve bandwidth. Ideal for sites with limited internet or high call concurrency.

Off

No negotiation — RingQ accepts whatever codec the remote side proposes. For testing and diagnostics only. Not recommended for production.

2.2 Transcoding

Transcoding is real-time audio conversion between different codec formats. When enabled (toggle shown active in the screenshot), RingQ acts as a media bridge — decoding the inbound audio stream and re-encoding it in the codec preferred by the other leg of the call.

✓ How transcoding works — example

SIP trunk → RingQ: Trunk sends audio encoded as PCMU (G.711 µ-law).

RingQ decodes the PCMU stream and re-encodes it as G.722 (wideband HD).

RingQ → IP Phone: Phone receives high-quality G.722 audio.

Result: Both legs operate at their best supported quality even though they use different codecs.

Enable Transcoding when

  • Your SIP trunk sends PCMU or PCMA but your phones support G.722 or OPUS for HD voice.
  • You have a mixed device environment with varying codec capabilities.
  • You want to normalise all internal calls to a single high-quality codec regardless of the trunk codec.

Disable Transcoding when

  • All trunks and endpoints share the same codec — passthrough uses far less CPU.
  • You have high call volume and limited server CPU — transcoding is computationally intensive.
  • You use Fax (T.38) or modem calls — transcoding corrupts fax signals.

2.3 Codec Priority List

The codec tiles in the Codec tab represent the active codecs offered by RingQ during SIP negotiation. Codecs listed first are offered with higher priority. RingQ supports the following codecs out of the box:

Codec

Type

Bitrate

Typical Use Case

PCMA

Narrowband

64 kbps

Standard telephony — Europe & Asia SIP trunks (G.711 a-law)

PCMU

Narrowband

64 kbps

Standard telephony — North America SIP trunks (G.711 µ-law)

iLBC

Narrowband

13–15 kbps

Poor network conditions — resilient to packet loss

G.722

Wideband

64 kbps

HD voice for internal calls between compatible IP phones

G.729

Narrowband

8 kbps

Low-bandwidth links with high call concurrency

OPUS

Fullband / Wideband

6–510 kbps

WebRTC, softphones, browsers — modern adaptive quality

2.4 Adding a Custom Codec

Use the Codec Name dropdown and +Add button at the top of the codec section to add a codec not shown in the default list.

  • Select the desired codec name from the Codec Name dropdown.
  • Click +Add to append it to the priority list.
  • Use the Arrange button to set its position in the priority order.

⚠ Note

Only add codecs supported by both your SIP trunk provider and your endpoint devices. Adding unsupported codecs will not cause failures but generates unnecessary SDP offer lines.

2.5 Arranging Codec Priority

Click the Arrange button (top-right of the codec grid) to enter drag-and-drop reorder mode. Drag codec tiles into your preferred order, then save.

Recommended priority order for most deployments

1. OPUS — Best quality for WebRTC softphones and modern devices.

2. G.722 — Wideband HD voice for compatible IP phones.

3. PCMA / PCMU — Universal compatibility with all SIP trunk providers.

4. G.729 — Low-bandwidth fallback for constrained links.

5. iLBC — Packet-loss resilient; useful for poor network conditions.

3. Saving & Applying Changes

  • Click Save to write the configuration to the RingQ database.
  • Click Cancel to discard all changes made in the current session.
  • A service restart may be required for network interface changes — RingQ will prompt you if needed.

⚠ Service Impact

Changing the Network Card Interface or External IP will affect all active calls. Schedule network changes during a maintenance window to avoid disrupting ongoing calls.

4. Troubleshooting

Symptom

Likely Cause & Fix

One-way audio on external calls

External IP is blank or set to the private NIC address. Enter the correct public / WAN IP in the External IP field.

SIP trunk registration fails

Wrong NIC selected. Ensure the selected interface has connectivity to the trunk provider’s network.

Poor call quality / audio distortion

Codec mismatch without transcoding active. Enable Transcoding and verify the codec priority list matches the trunk.

High CPU on the PBX server

Transcoding active for all calls. If all devices share a common codec, disable transcoding to allow passthrough.

488 Not Acceptable Here error

Strict negotiation mode with no matching codec. Switch to Adaptive or add the trunk’s required codec to the list.

Fax or modem calls fail or corrupt

Transcoding is re-encoding fax tones. Disable transcoding or use a dedicated SIP profile with T.38 pass-through.

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