Samsung iDCS 500 General Description Manual page 63

VOICE MAIL – INBAND INTEGRATION
The iDCS 500 system uses DTMF tones (inband signaling) to communicate with
any compatible voice mail system. Stations can call forward to a voice mail sys-
tem. When answered, the system will send DTMF tones routing the caller directly to
the called station user's mailbox. Keyset users can press one button to retrieve
messages from the voice mail system. A Voice Mail Transfer key permits keyset
users to easily transfer a caller directly to an individual voicemail box without navi-
gating through menus.
NOTE: Although most voice mail systems will work with the iDCS 500, the system
data has default values set to work with the Starmail Voice Processing System.
They may need to be changed if you are using another system.
VoIP
The iDCS 500 ITM3 VoIP card supports up to eight voice calls over an IP network
connection using the industry standards based H.323 protocol. An additional eight
VoIP channels can be added by installing an eight-circuit daughterboard for a total
of sixteen channels of VoIP . The ITM3 cards fit into any universal iDCS 500 card
slot. The iDCS 500 supports a maximum of two ITM3 cards per cabinet.
VoIP is transported by the iDCS 500 ITM3 card utilizing the ITU standards based
H.323 protocol. This standard addresses the means of transferring voice, data,
and images through IP (Internet Protocol) networks.
With VoIP certain compression standards have also been adopted to represent
each second of voice with an amount of bandwidth. The iDCS 500 ITM3 utilizes
G.711, G.729A or G.723 standards voice compression codec's. This allows for a
selectable 64kbps, 8Kbps or 6.3Kbps bandwidth use when preparing voice com-
pression for IP transport. Compression is used to reduce the digitized voice into a
smaller bandwidth that can be carried in smaller packets. The ITM3 H.323 gate-
way determines the compression method for each call setup. There is also a cer-
tain amount of frame/packet overhead in each compression channel. 64K of band-
width can support 6~7 calls simultaneously. This can vary depending on efficiency
features like Silence Suppression and multiframe counts. Unlike switched networks,
VoIP connections consist of a sequence of numbered of data packets. Since voice
conversation is usually considered "real time" these packets need to be delivered
in a consistent manner with minimal delay. This can be controlled via a Gatekeeper
which tracks and monitors voice packets. Gatekeepers are part of the H.323 stan-
dard but are not required. The iDCS 500 ITM3 is Gatekeeper compliant.
In any Ethernet environment, packet transfers are subject to delays and/or loss. If
these delays are greater than 200ms the voice quality will deteriorate. The Ethernet
data traffic and network topology should be a consideration when applying the
iDCS 500 ITM3 VoIP feature. Network congestion does affect call quality in any
VoIP application.
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