Tieline Bridge-IT, User Manual

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© Tieline Research Pty. Ltd. 2019
Bridge-IT Manual v4.0
17.6 Configuring the Jitter Buffer
Jitter, (also known as latency or delay), is the amount of time it takes for a packet of data to get
from one point to another. A jitter buffer is a temporary storage buffer used to capture incoming
data packets. It is used in packet-based networks to ensure the continuity of audio streams by
smoothing out packet arrival times during periods of network congestion. Data packets travel
independently and arrival times can vary greatly depending on network congestion and the type of
network used, i.e. LAN versus wireless networks. The concept of jitter buffering is displayed
visually in the following image.
Jitter buffer management is encompassed within Tieline's SmartStream IP technology which can:
· Remove duplicate packets.
· Re-order packets if they arrive out-of-order.
· Repair the stream in the event of packet loss (error concealment).
·
Manage delay dynamically based on current network congestion.
·
Manage forward error correction (FEC).
Tieline codecs can be used to program either a fixed or automatic jitter buffer and the setting you
use depends on the IP network you are connecting over. Over LANs, WANs and wireless
networks the automatic jitter buffer generally works well. It adapts automatically to the prevailing
IP network conditions to provide continuity of audio streaming and minimizes delay.
A fixed jitter buffer is preferable over satellite connections to ensure continuity of signals.
CAUTION: If a Tieline codec connects to a device that is using non-compliant RTP
streams then the last fixed setting entered into the codec will be enabled (default is
500ms). Non-compliant devices include some other brands of codec, web streams and
other devices.
Tieline ‘Auto Jitter Buffer’ Settings
The following automatic jitter buffer settings range from the most aggressive "Least Delay" setting,
which endeavors to minimize delay as much as possible while adapting to prevailing network
conditions, to the "Highest Quality" setting, which is the most conservative delay setting to
minimize packet loss when packet latency is not as critical. Best compromise is the most popular
setting used by most broadcasters as it achieves an excellent balance that minimizes latency for
bidirectional communications, as well as minimizing packet loss over most wired and wireless IP
networks.
Least Delay: This setting attempts to reduce the jitter buffer to the lowest possible point, while still
trying to capture the majority of data packets and keep audio quality at a high level. This setting is
the most aggressive in adapting to prevailing conditions, so the jitter buffer may vary more quickly
than with the other settings. It is not recommended in situations where jitter variation is significant,
or occurs in bursts. (E.g. cellular/multi-user wireless networks). It is best for stable and reliable
links such as dedicated or lightly-loaded WAN/LANs.