Understanding Video Conferencing Latency
Video conferencing makes business communication easier and more effective,
but only when the online experience is smooth and free of delays. The choppy
motion and lack of synchronization of some conferences stand in contrast to the
synchronization of high-quality conferences available via Nefsis.
What is the difference between a smooth video conference and a choppy,
jittery experience? The answer boils down to three factors:
- Bandwidth — the rate in bits per second at which
the network connection can move data in the conference (i.e., how wide is
the road from A to B?)
- Latency — the delay in milliseconds in the network
connection used by the conference (i.e., how smooth is the road and how much
other traffic is between A and B?)
- Routing — the network route that data packets
follow during the conference (i.e., how many turns does the road take between
A and B?)
When all of these factors are optimal, the video conference runs smoothly,
but all three factors are subject to variables such as other network traffic and
the status of equipment between the participant and the conference. Typically,
low bandwidth, high latency, and inefficient routing can lead to low-quality
Every video conference faces several technical realities:
- When a network connection is slow (i.e., shows high latency, low bandwidth
or poor routing), the computer sends fewer data packets across it.
- There is only one way out of the computer, and it's serial.
- Performance bottlenecks in the PC itself can affect video conference
- When packets of video conference data drop, the quality of the conference
decreases, and the process of resending the packets is not always error-free.
Nefsis deals with each of these realities in unique ways to ensure
high-quality video, voice and data in every conference.
Latency, bandwidth and routing
directly related to quality in the video conferencing experience. When latency
is low (e.g., 10-20ms), the connection between the computers will accept more
packets of data at a time, so the computer can send more and the quality of the
video conference will improve. Conversely, when latency is high (e.g., 120-180
milliseconds or more), the network connection accepts fewer packets at a time
and quality suffers.
The computer’s capacity for buffering data before sending it depends on the
TCP window size. By increasing and decreasing the TCP window size, computers
mutually ensure that each can transmit data at the optimal speed for the other.
However, even optimal window size cannot overcome extreme latency if there is
insufficient bandwidth or routing problems. Network congestion causes the buffer
to fill, so the video conferencing application has no choice but to drop some
data packets and send fewer across the network, resulting in lower quality.
Most applications simply rely on the operating system to control the TCP
window size, but as part of its Real-Time Routing Engine, Nefsis can control
this setting. To deal with latency, low bandwidth and routing delays, Nefsis
automatically optimizes data flow and compensates for delays in the network.
This includes using variable bitrate video encoding, also called dynamic
scalable video, which automatically
adjusts video quality, in real-time, on a per-connection basis, when bandwidth
availability does not permit desired settings.
By default, computers send data over the network serially. At any given time,
multiple processes are sending packets and multiple processes are waiting for
the packets at the other end, but as described above, some processes require
more time than others before their packets are ready to go to the network. On
any network, packets sent in 1-2-3-4-5-6 order may arrive in a different
sequence, like 1-4-2-5-3-6. This is not much of a problem in bursty,
asynchronous communication like e-mail or Web browsing, but out-of-sequence
packets can result in poor quality in the sustained, synchronous context of
Still, because networks are set up to move data serially, it is common for
video conferencing products to follow that model and push their packets to the
network serially, favoring throughput over conference quality. This presents
another problem, because the TCP window can take only so many packets, and
handing them off serially can lead to a buffer overflow. To prevent this
overflow, most video conferencing products fail, or throttle back and send
fewer packets, but that leads to the quality and performance problems
Serial processing requires that the source computer drop packets to maintain
synchronization or delay them to control the amount of data going to the
network; otherwise, there is the risk of overflowing buffers. The common remedy
is to reduce the amount of data going to the network, but that lowers the
quality of the online meeting.
Whereas most products maintain a serial connection during an online
meeting, Nefsis maintains a parallel connection process at the socket layer.
The parallel communication engine identifies packets at the source computer
and reorders them at destination to achieve the highest bandwidth possible.
Nefsis also controls the amount of data sent to the network based on available
At the ends of the video conference, participants’ computers play a role
in the quality of the online meeting. Factors such as the camera, the device driver
used with the camera, operating system updates, and software (especially
anti-malware) running in the background can affect the machine’s ability to
process video conferencing data, both sending and receiving.
Conditions on each participant’s LAN also come to bear on how the online
meeting performs. Proxy servers and firewalls, if too restrictive or improperly
configured, may create bottlenecks.
cloud-based, Virtual Conferencing Server (VCS) identifies any high-latency
connections in an online meeting and automatically compensates for the latency
in the flow of conference data to the participant’s computer.
Inevitably, some packets drop in the transmission of video conferencing data.
This manifests itself in missing regions of the on-screen image, distortion in
audio or video, and blank spaces in the session.
Because most traditional video conferencing products use UDP, which does not
provide for resending dropped packets, poor network conditions can severely
impact quality. Web conferencing products take advantage of common network
protocols for resending packets, but in greatly degraded network conditions,
some of these packets still remain irretrievable.
Nefsis’ VCS uses the parallel communication engine to act as a
broker by moving data packets among conference participants. The VCS also
acts as a real-time router to request re-send of dropped packets, receive
them, and forward them to all participants’ machines, which reorder
and reassemble them. The VCS does this not only on the current conference,
but for all conferences taking place on the server at any time.
Nefsis Diagnostic tools
Because the network plays such a prominent role in video conference quality,
Nefsis has developed tools for gauging its effects.
(These tools are for measurement only; they do not allow for configuration or
changing settings. For more details on how to use them, see the Bandwidth
Monitor and Network Diagnostics sections of the
Nefsis Online User Guide.)
Nefsis includes a Bandwidth Monitor utility, as shown in the example below.
Presenters and hosts can run the utility to measure total network bandwidth
usage during an online meeting. The utility describes the portion of bandwidth
dedicated to each of the processes at work in the online meeting: video, audio,
document sharing, desktop sharing and media file sharing.
- Bandwidth — Tests the available network capacity upstream (capacity to
send conference data over the current computer’s network connection) and
downstream (capacity to receive conference data over the connection). Higher
bandwidth generally implies higher-quality video conferences. In the example
below, Nefsis measures upstream bandwidth at 2035Kbps and downstream bandwidth
at over 14Mbps, well in excess of the recommended 1540 and 4096Kbps
- Latency — Tests the average delay in milliseconds in moving a packet
round-trip between the computer and the Nefsis Virtual Conference Server in
use for the current conference. The utility reports any packets that may drop
during the trip and the average response time for packets, as shown in the
- Routing — Describes and times the route across the Internet between the
computer and the Nefsis Virtual Conference Server. This test can identify
bottlenecks and slow connections along the route, as shown in yellow in the
Nefsis maintains multiple parallel processes for multiple streams of data —
VoIP, video, data, image capture, status — going over a single network
connection. It also manages the handoff of data to the network to keep the
quality of the video conference as high as possible for all participants.
Take a Free Trial or See a Live Demo
can take advantage of Nefsis' free trial and run all the network tests described above
on your own network. You can also contact us to schedule
a live demo and we can show you all these diagnostics over the web and
answer any latency-related questions you may have.
Related IT Topics
Multipoint Video — How Nefsis uses cloud computing to
deliver multipoint HD video
Bandwidth & QoS — Video conferencing bandwidth consumption
and QoS controls
Traffic Routing — Nefsis cloud and server configuration
options, showing traffic routing
Secure Video Conferencing — More details on SSL/TLS
encryption, certificates, and PKI