If there is excessive breakup of voice due to jitter with the default playout delay settings, increase the delay times. If
your network is small and jitter is minimal, decrease the delay times to reduce delay.
Before Cisco IOS Release 12.1(5)T, the playout-delay command was configured in voice-port configuration mode. For Cisco IOS Release 12.1(5)T and later releases, in most cases
playout delay should be configured in dial-peer configuration mode on the Voice over IP (VoIP) dial peer that is on the receiving
end of the voice traffic that is to be buffered. This dial peer senses network conditions and relays them to the DSPs, which
adjust the jitter buffer as necessary. When multiple applications are configured on the gateway, playout delay should be configured
in dial-peer configuration mode. When there are numerous dial peers to configure, it might be simpler to configure playout
delay on a voice port. If conflicting playout-delay values have been configured on a voice port and on a dial peer, the dial-peer
configuration takes precedence.
Playout delay is the amount of time that elapses between the time at which a voice packet is received at the jitter buffer
on the DSP and the time at which it is played out to the codec. In most networks with normal jitter conditions, the defaults
are adequate and you will not need to configure the playout-delay command.
In situations in which you want to improve voice quality by reducing jitter or you want to reduce network delay, you can configure
playout-delay parameters. The parameters are slightly different for each of the two playout-delay modes, adaptive and fixed
(see the playout-delay mode command).
In adaptive mode, the average delay for voice packets varies depending on the amount of interarrival variation that packets
have as the call progresses. The jitter buffer grows and shrinks to compensate for jitter and to keep voice packets playing
out smoothly, within the maximum and minimum limits that have been configured. The maximum limit establishes the highest value
to which the adaptive delay will be set. The minimum limit is the low-end threshold for incoming packet delay that is created
by the adaptive jitter buffer. Algorithms in the DSPs that control the growth and shrinkage of the jitter buffer are weighted
toward the improvement of voice quality at the expense of network delay: jitter buffer size increases rapidly in response
to spikes in network transmissions and decreases slowly in response to reduced congestion.
In fixed mode, the nominal value is the amount of playout delay applied at the beginning of a call by the jitter buffer in
the gateway and is also the maximum size of the jitter buffer throughout the call.
As a general rule, if there is excessive breakup of voice due to jitter with the default playout-delay settings, increase
playout-delay times. If your network is small and jitter is minimal, decrease playout-delay times for a smaller overall delay.
When there is bursty jitter in the network, voice quality can be degraded even though the jitter buffer is actually adjusting
the playout delay correctly. The constant readjustment of playout delay to erratic network conditions causes voice quality
problems that are usually alleviated by increasing the minimum playout-delay value in adaptive mode or by increasing the nominal
delay for fixed mode.
The minimum limit for playout delay is configured using the playout-delay (dial peer) command.
Use the show call active voice command to display the current delay, as well as high- and low-water marks for delay during a call. Other fields that can
help determine the size of a jitter problem are GapFillWith..., ReceiveDelay, LostPackets, EarlyPackets, and LatePackets.
The following is sample output from the show call active voice command:
ConnectionId[0xECDE2E7B 0xF46A003F 0x0 0x47070A4]
IncomingConnectionId[0xECDE2E7B 0xF46A003F 0x0 0x47070A4]
Separate H245 Connection=FALSE