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, FQDN Connections, and 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
- Sign up or log in to your Telnyx Mission Control Portal account
- Obtain your API Key
- Update your SIP connection via API with the jitter buffer settings
- Test call quality with your new configuration
Important Notes
- Jitter buffer is disabled by default — you must explicitly enable it
jitterbuffer_msec_mincannot exceedjitterbuffer_msec_max