VMware Horizon
Real-Time Audio-Video (RTAV) Feature Explained
With the Real-Time Audio-Video feature, you can use the
local client system's webcam or microphone in a remote desktop or published
application. Real-Time Audio-Video is compatible with standard conferencing
applications and browser-based video applications. It supports standard
webcams, audio USB devices, and analog audio input.
End users can run Skype, Webex, Google Hangouts, and other
online conferencing applications in their remote desktops. This feature
redirects video and audio data to the agent machine with a lower bandwidth than
can be achieved by using USB redirection. With Real-Time Audio-Video, webcam
images and audio input are encoded on the client system and then sent to the
agent machine. On the agent machine, a virtual webcam and virtual microphone
can decode and play the stream, which the third-party application can use.
No special configuration is necessary, although
administrators can set agent-side group policies and registry keys to configure
frame rate and image resolution, or to turn off the feature. By default, the
resolution is 320 by 240 pixels at 15 frames per second. If needed,
administrators can also use client-side configuration settings to set a
preferred webcam or audio device.
Note This feature is available only on some types of
clients. To find out whether this feature is supported on a particular type of
client, see the feature support matrix included in the installation and setup
document for the specific type of desktop or mobile client device.
Real-Time
Audio-Video (RTAV) Design
Architectural
Challenges of Running Real-Time Audio and Video in a VDI Environment
Previously, VoIP and video support in VDI environments was
limited by architectural issues that prevented audio and video conferencing
from working optimally in a virtual machine. These issues included:
·
Heavy CPU load on data center servers – All
processing for VoIP and video chat calls was handled on the data center
servers.
·
Media hair-pinning – VoIP and
videoconferencing traffic was not sent point-to-point but streamed through the
data center network and server.
·
High bandwidth usage – Audio and video
traffic was not encoded with standardized codecs but was sent as raw USB
traffic, resulting in extremely high bandwidth usage.
·
No quality of service (QoS) – Audio and
video traffic was sent inside the display protocol, which did not provide
granular QoS policies to prioritize VoIP and videoconferencing traffic.
