Polycom SIP 3.1 Low-Delay Audio Packet Transmission, Jitter Buffer and Packet Error Concealment

Models: SIP 3.1

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•Dynamic Noise Reduction

Administrator’s Guide SoundPoint IP / SoundStation IP

Dynamic Noise Reduction

Treble/Bass Controls

Low-Delay Audio Packet Transmission

The phone is designed to minimize latency for audio packet transmission.

There are no related configuration changes.

Jitter Buffer and Packet Error Concealment

The phone employs a high-performance jitter buffer and packet error concealment system designed to mitigate packet inter-arrival jitter and out-of-order or lost (lost or excessively delayed by the network) packets. The jitter buffer is adaptive and configurable for different network environments. When packets are lost, a concealment algorithm minimizes the resulting negative audio consequences.

Configuration changes can performed centrally at the boot server or locally:

Central

Configuration file:

Set the jitter buffer tuning parameters including minimum and

(boot server)

sip.cfg

maximum size and shrink aggression.

 

 

For more information, refer to Codec Profiles <audioProfile/> on

 

 

page A-41.

 

 

 

Local

Web Server

Set the jitter buffer tuning parameters including minimum and

 

(if enabled)

maximum size and shrink aggression.

 

Navigate to http://<phoneIPAddress>/coreConf.htm#au

 

 

 

 

Changes are saved to local flash and backed up to <Ethernet

 

 

address>-phone.cfgon the boot server. Changes will permanently

 

 

override global settings unless deleted through the Reset Local

 

 

Config menu selection and the <Ethernet address>-phone.cfgis

 

 

removed from the boot server.

 

 

 

Voice Activity Detection

The purpose of voice activity detection (VAD) is to conserve network bandwidth by detecting periods of relative “silence” in the transmit data path and replacing that silence efficiently with special packets that indicate silence is occurring. For those compression algorithms without an inherent VAD function, such as G.711, the phone is compatible with the comprehensive codec-independent comfort noise transmission algorithm specified in RFC 3389. This algorithm is derived from G.711 Appendix II, which defines a comfort noise (CN) payload format (or bit-stream) for G.711 use in packet-based, multimedia communication systems. The phone generates CN

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Polycom SIP 3.1 Low-Delay Audio Packet Transmission, Jitter Buffer and Packet Error Concealment, Voice Activity Detection