# Configure Jitter Buffer

A jitter buffer holds incoming audio packets for a fraction of a second before playing them back. That short pause gives your phone system time to reorder packets that arrive late or out of sequence, which smooths out choppy audio. "Jitter" is the variation in how evenly voice packets arrive across the network: when that timing gets uneven, calls start to sound broken up. Turn the buffer on when call quality suffers from network problems.

:::note
PBX stands for Private Branch Exchange, which is the phone system that routes your calls. In this guide the PBX is your Cloud Voice service.
:::

## When to use it

If calls are affected by [packet loss](/pbx/administrator-guide/jitter-buffer-overview/#what-is-jitter-buffer__packet-loss) or [packets arriving out of order](/pbx/administrator-guide/jitter-buffer-overview/#what-is-jitter-buffer__packets-out-of-order), enabling the jitter buffer can smooth out the audio and improve the overall call experience.

## Configure the jitter buffer

1. Sign in to the Cloud Voice management portal and go to **PBX Settings > Jitter Buffer**.

2. Turn on **Jitter Buffer**.

   :::caution
   Turning on the master **Jitter Buffer** switch does not smooth any audio on its own. Buffering only applies to the specific trunks and extensions you move into the **Selected** box in the next steps. If you skip that, the feature stays inactive.
   :::

3. To apply jitter buffering to trunks, move the trunks you want from the **Available** box into the **Selected** box.

   Outbound audio sent over a selected trunk is dejittered at the far end.

   :::note
   A trunk is the connection between your phone system and the outside phone network (your voice provider, reached over a SIP, or Session Initiation Protocol, link). An extension is an internal user or device, such as a desk phone or the Cloud Voice App. "Dejittered" simply means the buffer has smoothed the audio before it is played.
   :::

4. To apply jitter buffering to extensions, move the extensions you want from the **Available** box into the **Selected** box.

   Audio received on a selected extension is dejittered before playback.

5. From the **Implementation** drop-down list, choose how the jitter buffer behaves:

   - **Adaptive**: The buffer resizes itself to match current network delay, so the delay added to packets leaving the buffer varies. When you select this option, set the following:
     - **Adaptive Adjustment Size (ms)**: How much the buffer grows with each adjustment. The default is 50. With the default in place, the buffer starts at 0 ms and increases in 50 ms steps as network conditions demand.
     - **Max Jitter Buffer Size (ms)**: The largest size the adaptive buffer is allowed to reach. The default is 200.
   - **Fixed**: The buffer keeps a constant size, so every packet leaving it carries the same delay. When you select this option, enter a value in the **Jitter Buffer Size (ms)** field. The default is 200.

   :::tip
   Leave these values at their defaults unless you have a specific reason to change them. **Adaptive** fits most networks because it adjusts to changing conditions on its own. Choose **Fixed** when you want a predictable, unchanging amount of delay.
   :::

   :::caution
   A larger buffer can absorb more jitter, but it also holds each packet longer, which adds delay (latency) to the call. Setting the size too high can make a conversation feel laggy, with people talking over each other. Raise it only as far as you need to clear the audio problem.
   :::

6. Click **Save**, then **Apply**.

   :::note
   Your changes are not live until you click **Apply**. **Save** stores the settings, and **Apply** pushes them to the running phone system.
   :::