/mcpThe information presented in this document was created from devices in a specific lab environment. All of the devices used in this document started with a cleared (default) configuration. If you are working in a live network, ensure that you understand the potential impact of any command before using it. Conventions For more information on document conventions, refer to the Cisco Technical Tips Conventions. Causes of Choppy Voice Choppy voice quality is caused by voice packets being either variably delayed or lost in the network. When a voice packet is delayed in reaching its destination, the destination gateway has a loss of real−time information. In this event, the destination gateway must predict what the content of the missed packet can possibly be. The prediction leads to the received voice not having the same characteristics as the transmitted voice. This leads to a received voice that sounds robotic. If a voice packet is delayed beyond the prediction capability of a receiving gateway, the gateway leaves the real−time gap empty. With nothing to fill up that gap at the receiving end, part of the transmitted speech is lost. This results in choppy voice. Many of the choppy voice issues are resolved by making sure that the voice packets are not very delayed (and more than that, not variably delayed). Sometimes, voice activity detection (VAD) adds front−end clipping to a voice conversation. This is another cause of choppy (or clipped) voice. The various sections in this document show how to minimize the instance of choppy voice. Most of these measures require assuring the introduction of minimum jitter in your voice network. Bandwidth Requirement Before you consider applying any measures for minimizing jitter, provision sufficient network bandwidth to support real−time voice traffic. For example, an 80 kbps G.711 VoIP call (64 kbps payload + 16 kbps header) sounds poor over a 64 kbps link because at least 16 kbps of the packets (which is 20 percent) are dropped. The bandwidth requirements vary based on the codec used for compression. Different codecs have different payloads and header requirements. Usage of VAD also affects the bandwidth requirement. If Real Time Protocol (RTP) header compression (cRTP) is used, it can further lower the bandwidth requirement. For example, the bandwidth required for a voice call using the G.729 codec (default 20 byte payload) with cRTP, is like this: Voice payload + compressed (RTP header + User Datagram Protocol (UDP) header + IP header) +Layer 2 header • This is equivalent to: 20 bytes + compressed (12 bytes + 8 bytes + 20 bytes) + 4 bytes • This equals: 28 bytes, since the header compression reduces the IP RTP header to a maximum of 4 bytes. This yields 11.2 kbps at an 8 kbps codec rate (50 packets per second). • For more information, refer to Voice over IP − Per Call Bandwidth Consumption.