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Business Telephones

Telecommunications Consulting
 
Call Barring
Caller Display
Call Intrusion/Call Pickup
Call Waiting
Hot Desking
Parking Calls
Ring Back When Free/ Callback
Message Waiting Indication
Ring Tones
Music on Hold (MOH)
How the System Receives Time
The 'No User' User
DND, Follow Me and Forwarding
Do Not Disturb (DND)
Follow Me
Forward Unconditional
Forward on Busy
Forward on No Answer Determining a User's Busy Status
Chaining and Loops
Transferring Calls
Off-Switch Transfer Restrictions
Overview of Hunt Groups
Hunt Group Types
Call Presentation
Hunt Group Member Availability
Using Queuing
Hunt Group Overflow
Fallback
Hunt Group Voicemail
Hunt Group Examples
Conferencing Overview
Short Codes
Short Code Fields and Characters
User Dialing
Application Dialing-Secondary Dial Tone
Line Short Codes
Least Cost Routing Overview
Short Code Matching Examples
Short Code List
Overview of Data Routing
NAT & DHCP
Establish ISDN Internet Connection
ISDN Link Between IP Offices
Using a Dedicated T1/PRI ISP Link
Direct Remote Access
Creating a VoIP Link via the WAN Port Using PPP
Create VoIP Link via WAN Port Using Frame Relay
Overview of VoIP
VoIP Protocols-Performance
Small Community Networking
Enabling Small Community Networking
SCN Short Codes
Connecting a Transactional Pad
Paging / Universal Paging Access Module
Paging via an Analog Extension Port
Paging via an Analog Extension Trunk
Making Page Calls
Paging Via Voicemail Pro
Dial By Name
Using Dial by Name
Country Locales
VoIP Implementation
 
Hot Desking
 
IP Office Manager Pt.1

IP Office Manager Pt.2

 

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Avaya IP Office

The Avaya IP Office platform is the ultimate in converged voice and data technology. IP Office brings a combination of voice and data applications formerly reserved for only the largest corporations. Cutting edge customer service with easy to use tools is now available to the smallest of businesses.

 

 

   
   

 

Overview of VoIP

Depending on the model of control unit the IP Office control unit supports up to 60 voice compression channels. Either 3 or 16 of these channels are pre-installed in Small Office Edition controls units. On other units they are added by installing Voice Compression Modules (VCM's). The type and number of VCM modules supported by each control unit type varies.

The voice compression channel improves call quality and can be used to compress voice down to either 6k3 (G723) or 8k (G729/Netcoder) and provides echo cancellation (required for high latency circuits).

The bandwidth required for a VoIP call is made up of two parts, one of which is due to the actual digitization of the analog voice the other is required by the protocol which is used to wrap the digitized voice up and transport it to the remote site. VoIP calls require an overhead of 40 bytes per packet (RTP/UDP/IP Header overhead) this overhead is increased on a LAN by a further 12 bytes Ethernet or by 7 bytes over a PPP WAN link.

When transporting voice over low speed links (WANs) it is possible that normal data packets (typically 1500 byte IP packets) can prevent or delay the voice data from getting across the link. This can cause a very unacceptable speech quality. Thus it is vital that the routers in the network that carry voice have some form of Quality of service mechanism (QoS).

The Control Unit supports the DiffServ (RFC 2474) Quality of Service mechanisms (QoS) which is based upon a Type of Service (ToS) field in the IP header. The software will prioritize voice, fragment large packets and provide VoIP header compression to minimize the WAN overhead.

Typically the VoIP WAN overhead is 47 bytes on 20 byte payload this is 235% overhead. On the WAN protocol this is reduced to 11 bytes (8 bytes data, 2 bytes CRC and 1 byte HDLC flag) on the same 20-byte packet this is only 55%, and 180% saving. This overhead must be included when calculating the actual link speeds required to support voice traffic. For example an 8Kbps compression voice path actually required 12.4Kbps of WAN bandwidth when using QoS or 26.8Kbps if using standard non QoS routers.

QoS routers are also required to ensure low speech latency and to maintain sufficient audible quality. At present our header compression is based upon the latest standards (RFC 2507/2508/2509). For efficiency we operate below PPP (non-standard) - reducing the overheads further and allow data fragmentation to be performed more effectively (keeping latency low). It is therefore required to place our equipment at both ends to operate at full efficiency.

Depending on the model of control unit the IP Office control unit supports up to 60 voice compression channels. Either 3 or 16 of these channels are pre-installed in Small Office Edition controls units. On other units they are added by installing Voice Compression Modules (VCM's). The type and number of VCM modules supported by each control unit type varies.

The voice compression channel improves call quality and can be used to compress voice down to either 6k3 (G723) or 8k (G729/Netcoder) and provides echo cancellation (required for high latency circuits).

The bandwidth required for a VoIP call is made up of two parts, one of which is due to the actual digitization of the analog voice the other is required by the protocol which is used to wrap the digitized voice up and transport it to the remote site. VoIP calls require an overhead of 40 bytes per packet (RTP/UDP/IP Header overhead) this overhead is increased on a LAN by a further 12 bytes Ethernet or by 7 bytes over a PPP WAN link.

When transporting voice over low speed links (WANs) it is possible that normal data packets (typically 1500 byte IP packets) can prevent or delay the voice data from getting across the link. This can cause a very unacceptable speech quality. Thus it is vital that the routers in the network that carry voice have some form of Quality of service mechanism (QoS).

The Control Unit supports the DiffServ (RFC 2474) Quality of Service mechanisms (QoS) which is based upon a Type of Service (ToS) field in the IP header. The software will prioritize voice, fragment large packets and provide VoIP header compression to minimize the WAN overhead.

Typically the VoIP WAN overhead is 47 bytes on 20 byte payload this is 235% overhead. On the WAN protocol this is reduced to 11 bytes (8 bytes data, 2 bytes CRC and 1 byte HDLC flag) on the same 20-byte packet this is only 55%, and 180% saving. This overhead must be included when calculating the actual link speeds required to support voice traffic. For example an 8Kbps compression voice path actually required 12.4Kbps of WAN bandwidth when using QoS or 26.8Kbps if using standard non QoS routers.

QoS routers are also required to ensure low speech latency and to maintain sufficient audible quality. At present our header compression is based upon the latest standards (RFC 2507/2508/2509). For efficiency we operate below PPP (non-standard) - reducing the overheads further and allow data fragmentation to be performed more effectively (keeping latency low). It is therefore required to place our equipment at both ends to operate at full efficiency.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 
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