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.
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.
jitterbuffer_msec_min cannot exceed jitterbuffer_msec_max