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VOIP FAQ's   

Network Requirements

Quality of Service (QoS) is a measure of the performance of a network that reflects the availability of network service and the quality of network transmissions. The term itself refers to a number of networking technologies and techniques and does not necessarily restrict itself to any single protocol or standard.

There are a number of measures that can be taken on the LAN and WAN to make them 'good enough' to carry voice traffic. Some of these are the implementation of standards based QoS protocols while are simply a matter of network architecture and good network management practices.

The term 'good enough' is intentional. Every customer will have different expectations and different budgets to work to. Some will be willing to upgrade their networks to use the best possible equipment and practices. To others the additional expense may be viewed as unnecessary.

Examples of standards based Quality of Service protocols include DiffServ and 802.1p/q.
 

What are Voice Compression Modules (VCM's) for?

VCM's are required to support the following scenarios:

Usage of Embedded Voicemail on the Small Office Edition (used as a memory boost by compressing the voice files).
Internal phone calls between an IP device and a non-IP device.
Analog/digital phones to IP trunks (SIP/H.323) including managed Frame Relay and managed IP VPN (provides echo cancellation).
IP phones to ISDN or PSTN trunks (convert IP to TDM and vice-versa).

 

IP device to IP device
Call progress tones (for example dial tone, secondary dial tone, etc) do not require voice compression channels with the following exceptions:

Short code confirmation, ARS camp on and account code entry tones require a voice compression channel.
Devices using G.723 require a voice compression channel for all tones except call waiting.

 

When a call is connected:

If the IP devices use the same audio codec no voice compression channel is used.
If the devices use differing audio codecs, a voice compression channel is required for each.

 

“Direct Media” is a VoIP concept that circumvents resources (TDM bus, Gateway) within the system and improves the voice quality. If two IP devices are connected on the same system, a direct LAN connection between them will be established once the call has been set up (as long as they use the same Codecs).

It is possible for an IP device to be physically located at one site while being registered at a different site. In this case, even for VoIP across the WAN the VCM would not be used, as long as the two IP devices involved in a phone call are registered on the same system.
 

Audio Codec

RTP Voice Data Payload

Packets per Second

LAN (bps)

% Overhead LAN

WAN (bps)

% Overhead WAN

Algorithmic Delay (milli-seconds)

G.723.1

24 Bytes

33.33

20,800

225%

9,867

54%

80

G.729a

20 Bytes

50

29,600

270%

13,200

65%

40

G.711 (64K)

160 Bytes

50

85,600

34%

69,200

8%

20

 
Note:  IP end-points do NOT require data channels
 
What is the Maximum Number of Simultaneous VoIP Calls?

Each IP Office can be fitted with an optional Voice Compression Module (VCM) to support VoIP connections.

The IP406 can be fitted with a single module offering up to 30 simultaneous calls.
The IP412 is capable of supporting two modules of all types, allowing up to 60 simultaneous calls.
The IP500 is capable of supporting two VCM 32/64 modules allowing up to 128 simultaneous calls.
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