---
title: Reduce Audio Jitter on SIP Connections | Telnyx Release Notes
canonical: https://telnyx.com/release-notes/adaptive-jitter-buffer-sip-connections
---
# Reduce Audio Jitter on SIP Connections | Telnyx Release Notes

[Release notes](https://telnyx.com/release-notes)

# Reduce Audio Jitter and Improve Call Quality on SIP Connections

### 16, Feb 2026

Telnyx SIP Connections now support adaptive jitter buffer configuration. Enable jitter buffering and define custom min/max buffer values to reduce audio artifacts and improve call quality for your specific network conditions.

**Who is this for:** Customers using Credential, FQDN, or IP-based SIP connections via API — especially contact centers, VoIP deployments, or any scenario where packet reordering impacts audio quality.

**Note:** This feature is currently API-only. Portal UI controls are coming soon.

## What's New

Three new connection settings are now available:

| Setting                 | Purpose                        | Default | Range  |
| ----------------------- | ------------------------------ | ------- | ------ |
| enable\_jitter\_buffer  | Toggle jitter buffering on/off | false   | —      |
| jitterbuffer\_msec\_min | Minimum buffer size (ms)       | 60      | 40–400 |
| jitterbuffer\_msec\_max | Maximum buffer size (ms)       | 200     | 40–400 |

These settings are available on [Credential Connections](https://developers.telnyx.com/api-reference/credential-connections/update-a-credential-connection), [FQDN Connections](https://developers.telnyx.com/api-reference/fqdn-connections), and [IP Connections](https://developers.telnyx.com/api-reference/ip-connections).

## Key Benefits

* **Improved audio quality:** Reduces audio artifacts caused by network jitter and packet reordering on RTP streams.
* **Customizable buffering:** Larger values add latency but tolerate more jitter; smaller values reduce latency but are more sensitive to network conditions.
* **Per-connection control:** Configure jitter buffer settings individually for each SIP connection based on your use case.

## Getting Started

1. Sign up or log in to your [Telnyx Mission Control Portal account](https://portal.telnyx.com/)
2. [Obtain your API Key](https://developers.telnyx.com/docs/development/authentication)
3. Update your SIP connection via API with the jitter buffer settings
4. Test call quality with your new configuration

## Important Notes

* Jitter buffer is disabled by default — you must explicitly enable it
* `jitterbuffer_msec_min` cannot exceed `jitterbuffer_msec_max`

[Start building](https://telnyx.com/sign-up)

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