In voice over IP (VoIP), jitter is the variation in the time between packets arriving, caused by network congestion, timing drift, or route changes. A jitter buffer can be used to handle jitter.
To make it easier to understand, you are converting analog voice into digital IP packets that are sent across the internet. The protocol used is UDP which does not resend lost or dropped packets therefore some packets may arrive out of order or wont arrive at all. The SIP endpoints being your Vonage telephone adapter and the Vonage PSTN Gateways which interconnect the IP packets to the PSTN phone networks have jitter buffers which attempt to correct out of order packets or lost packets arriving. However when the maximum jitter is too high, the jitter buffers are unable to buffer the packets in the correct order or they don't receive enough packets to buffer before sending to the analog side for playback. This means that either you, the party your speaking with, or both sides could hear choppy or garbled audio.
If your jitter is too high, you would need to first identify where you have high latency or packet loss. It could be on your own LAN, at your modem, your ISP, or at an IP hand off between your ISP and Vonage.
To better understand and help you with your issue, the folks in this forum will request you to run the test from the 2nd sticky at the top of the forum page which I am listing for you here. There will also be additional tests that you will have to run to identify where the problem occurs for you.
1. Your location 2. Your ISP name and type (cable, DSL, wireless etc) 3. Upload and download speeds 4. Modem make and model 5. Type of Vonage adapter used ie RT31P2 or PAP2 6. Setup of Network ie Modem---->Vonage Adapter-->PC 7. Issue you are experiencing ie dropped calls, choppy audio etc 8. Run this Voip Test: After the test has fully completed, in the Advanced Tab, click on "View Text" in the lower right hand corner. Copy-N-Paste that data into your post.
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