by April 1, 2009 0 comments

Be it a UC solution or some other enterprise app, one question at the back of
every CIO’s mind is how much bandwidth is enough? Couple of months back we spoke
about VoIP and the minimum bandwidth required for VoIP and role of Codecs in the
entire communication. On similar lines, this month we are going to look at
bandwidth requirements for video over IP. In video communications over IP there
are a lot of factors that make the difference between good and bad quality, for
eg quality of the camera, video codecs used and complexity of the scene. Here we
first talk a little about the most commonly used codecs in video communication
and then take a look at how can you measure bandwidth.

Lets start with the video codecs as these are often referred to as the
brains of voice as well as video communications. Here H.323 is the commonly used
video codec for multimedia conferences. H.323 was defined by ITU as a globally
accepted standard for multimedia communication over IP Networks.

This standard includes functionalities of its previous versions, H.321 and
H.320. This standard is supported by most of the video conferencing vendors. In
low bandwidth scenarios it uses G.723 codec for audio communication and H.263
for video communications. For data conferencing it specifies use of T.120
standard. H.323 has been long competing with another popularly used standard:
SIP. Both these protocols are widely supported by most of the video conferencing
devices. Next version of H.323, H.325 which is referred as next generation
multimedia system is expected to come out in 2010. A nice comparison between SIP
and H.323 can be found at

Codecs in OCS 2007
Microsoft uses codes RTAudio and RTvideo in its office communication server
2007. RT audio, also known as Real time Audio, is a speech codec for 2-way VoIP
applications. According to Microsoft, the decoder part of the codec has a jitter
control module which does active management of packet losses and jitter and also
has an error concealment module. Minimum bandwidth required for RTAudio is
24Kbps while max bandwidth supported is 74Kbps.

You can also manually set bandwidth used by Office Communication Server 2007.
To do this, you need to install communicator.adm security template. Here you
will find the list of OCS group policies. Look for MaxAudioVideoBitRate policy
option. This option is used to limit bandwidth used by audio as well as video
calls. The recommended setting for this is 345Kbps,i.e 300Kbps for video and
45Kbps for audio. If you are adventurous enough, you can also try to disable
this policy and measure how much bandwidth is used by OCS 2007.

Microsoft also provides a Quality of Experience Monitoring Server for Office
Communication Server 2007. With this tool you can monitor media quality and
collect statistics related to media quality. It also comes with diagnostics
tools which can bed used to diagnose VoIP related complaints. It collects
quality metrics after end of every VoIP call which it then stores in its SQL
database. This information is futher used by the sever to generate trends and
alert users as well as generate media quality reports. It can be downloaded from

Bandwidth for Tele-Presence
Speak to any Telepresence vendor about how much bandwidth is required for
life-like experience over telepresence and they will give answers varying from
1Mbps to 2 Mbps. Infact many vendors recommend 2Mbps for the best experience.
Thanks to the nice people at Tandberg we received a personal Tele-Presence
solution called tandberg MX 1700. We tested out the solution to see how much
minimum bandwidth it can operate at. While trying the solution we even managed
to connect the solution over 128 Kbps speed, but at this speed video was jittery
even though audio worked just fine. To conclude, at least 256 Kbps is needed to
have a conversation with this solution. However, for a decent real life like
conversation, speed of at least 1 Mbps is required.

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