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BCM 2.0 IP Telephony
Configuration manual
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Avaya BCM 2.0 IP Telephony Configuration manual

Enterprise Edge 2.0 IP Telephony Configuration Guide
1-800-4 NORTEL www.nortelnetworks.com © 2000 Nortel Networks
P0911590 Issue 02

Contents

Chapter 1 Overview 7
About this document 8 System Functions 8 Dialing plan support 9 Network Quality of Service 10
Network Monitoring 10 Quality of service parameters 10
Fallback to circuit-switched voice facilities 11 Network Performance Utilities 12 Codecs 12 Silence compression 13 Echo cancellation 13 Jitter buffer 14 Fax calls 14 Alarm Notification 14
Contents 3
Chapter 2 Engineering guidelines 15
Introduction 15
Enterprise Edge IP telephony 15 Overview 15 Enterprise Edge VoIP Gateway bandwidth engineering 17 Multiple network interfaces 18
Method 1 19
Method 2 19 LAN engineering 20
Silence compression 20 WAN engineering 22
Determining WAN link resources 23
Link utilization 23
Determining network loading caused by IP telephony traffic 24
Other intranet resource considerations 25 Setting QoS 26 Measuring Intranet QoS 27
Measuring end-to-end network delay 27
Measuring end-to-end packet loss 28
Recording routes 28
Adjusting ping measurements 29
Measurement procedure 29
Other measurement considerations 30 Further network analysis 31
4 Contents
Components of delay 31
Reduce link delay 32
Reducing hop count 32
Routing issues 34 Implementing QoS in IP networks 34
Traffic mix 35
TCP traffic behavior 35
Enterprise Edge Router QoS Support 35 Implementing the network 36
LAN engineering 36
Setting the Quality of Service threshold for fallback routing 36 IP telephony settings 37
Codec types 37
Silence compression 38
Echo cancellation 40
Jitter buffer 40
Fax calls 41
Fallback threshold 42 Dialing plan 43
IP telephony and M1 networking 43
Enterprise Edge and a Gatekeeper 46
Toll bypass with VoIP telephony 47
Core telephony services configuration 52 Post-installation network measurements 52
Setting IP telephony QoS objectives 53
Intranet QoS monitoring 53
User feedback 54
Chapter 3 Engineering checklist 55
Chapter 4 Installation 57
Installation flowchart 57
Configuring a local gateway 57
Configuring a remote gateway 58
Chapter 5 Configuration 61
User Interface Overview 62
Local Gateway Configuration 63
Remote Gateway Configuration 65
Core telephony services configuration 67
Configuration of fallback to conventional circuit-switched facilities 67
Chapter 6 Maintenance 69
Quality of Service Monitor 69
Quality of Service Status 69
Using the QoS Monitor pull-down View menu 69 Operational Statistics 69 Backup and Restore Procedures 69
Chapter 7 Interoperability 71
Interoperability considerations 71
Asymmetrical media channel negotiation 72
No feedback busy station 72
Glossary 73
Index 75
Contents 5
6 Contents

Overview

The Enterprise Edge VoIP Gateway reduces communication costs by routing voice traffic over private Internet Protocol (IP) networks as part of the Enterprise Edge product portfolio. Enterprise Edge uses IP telephony to link multiple sites together using an existing corporate data network. The IP trunks are an important part of the telephony services. IP telephony is transparent to users.
IP telephony involves the conversion of voice from its normal telephony format (continuous analog or digital signal) into a digital packet format transported over an intranet.
IP telephony operates on an installed corporate IP network. IP telephony requires a correctly managed intranet, instead of the interne t. The private IP network facilities must have sufficient bandwidth on the private Wide Area Network (WAN) backbone. The Engineering guidelines chapter on page 15 contains information about determining if your corporate IP network can support IP telephony. A keycode controls the number of supported IP ports.
IP telephony uses a Web-based browser for configuration. See the Configuration chapter on page 61 for information about how to configure IP telephony.
The VoIP Gateway supports ITU-H.323v2 gatekeeper operation. The VoIP Gateway allows Enterprise Edge systems to use a central facility for IP address resolution. The VoIP Gateway uses standard Digital Signal Processor (DSP) voice coding. The VoIP Gateway supports compression algorithms (codecs) such as G.711, G.723, and G.729. See Codec types on page 37 in the Engineering
guidelines chapter for information about codecs.
The VoIP Gateway monitors the data network and reroutes calls to the conventional circuit-switched voice facilities if Quality of Service (QoS) over the data network decreases. This Fallback to Conventional Circuit-Switched Voice Facilities feature allows the system and installer to determine the acceptable QoS over the data network. The customer can configure QoS parameters according to their requirements. See the Quality of service parameters section on page 10 and
Configuration of fallback to conventional circuit-switched facilities section on
page 67 in the Configuration chapter for information about configuring the QoS parameters. If the quality is below the expected level of QoS, the conventional circuit-switched voice facilities route is selected until the QoS returns to an acceptable level.
The VoIP Gateway also activates fallback for other cases when a call cannot be completed. Examples include no response from the destination, the remote gateway table is improperly configured, or if there are not enough DSP resources to support the call.
P0911590 Issue 02 Enterprise Edge 2.0 IP Telephony Configuration Guide
8 Overview

About this document

This guide provides information about the Enterprise Edge VoIP Gateway. This guide is for both guide telecom and datacom engineers who design and install the network. The assumption is that the telecom engineer understands how to engineer the Enterprise Edge product portfolio, and get system voice and fax traffic statistics. The assumption is that the datacom engineer understands the intranet architecture, LAN implementation, tools for collecting and analyzing data network statistics, and data network management systems. The terms installer and administrator used in this document refer to either the telecom or datacom engineer. This guide contains the following chapters:
• Engineering guidelines on page 15
• Engineering checklist on page 55
• Installation on page 57
• Configuration on page 61
• Maintenance on page 69
• Interoperability on page 71

System Functions

Enterprise Edge VoIP Gateway uses IP telephony to provide least cost routing of voice traffic through a corporate intranet. VoIP Gateway provides the following:
• Basic calls with answer and disconnect supervision
• Direct Inward Dial (DID) and Direct Outward Dial (DOD)
• Calling name and number
• VoIP Gateway to M1-ITG capability
• ITU-H.323v2 compatible gateway
• ITU-H.323v2 Gatekeeper interoperability
• Economical bandwidth use through voice compression
• Economical bandwidth use through silence compression
• Quality of Service (QoS) monitoring of gateways
• Circuit-switched voice facilities fallback capability The core telephony service made available through Enterprise Edge considers the
Enterprise Edge VoIP Gateway as a trunk. The IP trunk uses the trunking and routing functionality of the Enterprise Edge product portfolio. The IP trunks are an important part of the Enterprise Edge product portfolio.
Enterprise Edge 2.0 IP Telephony Configuration Guide P0911590 Issue 02
Overview 9
VoIP Gateway trunks, are supervised trunks with answer and disconnect supervision. The VoIP Gateway supports voice and fax calls. See the Engineering
guidelines chapter on page 15 for more information about fax calls. VoIP Gateway
does not support modem calls. The IP telephony gateway allows communication with other supported gateways
and H.323v2 gateways through trunk calls. The IP telephony gateway supports Direct Routed communication. The local gateway performs the address resolution and maintains the remote gateway table.
The ITU-H.323v2 Gatekeeper allows for centralized configuration of IP address information. Two call signaling protocols are available in Enterprise Edge 2.0.
• Direct routed, where Enterprise Edge uses a locally maintained table for IP resolution. The locally maintained table is the remote gateway table. See the
Configuration chapter on page 61 for information about the remote gateway
table.
• Gatekeeper routed, where Enterprise Edge uses a centralized gatekeeper for address resolution. In this mode, the gatekeeper handles all call control signaling. The remote gateway table is not used.

Dialing plan support

Dialing plan configuration allows the customer to set up the routing tables to route calls to appropriate destinations based on the dialed digits.
Routing codes and the destination code table allow the core telephony services on the Enterprise Edge to direct the use and time of use of trunking facilities for calls. Routing codes are associated with line pools. You can assign more than one routing code to each destination code, depending on factors such as Least Cost Routing.
Enterprise Edge has two main areas of configuration: the destination codes in the core telephony services and the destination digits in the remote gateway configuration table. The destination digits allow VoIP Gateway to route calls to the appropriate intranet destination based on the leading dialed digits. The destination code tables route calls to the appropriate trunks based on the leading dialed digits.
See the Configuration chapter on page 61 for details on configuring destination digits and destination codes.
The dialing plans for all VoIP Gateways connected to the corporate intranet require planning to allow calls between gateways as required.
For more information about Dialing plan support, see Dialing plan section on page 43.
P0911590 Issue 02 Enterprise Edge 2.0 IP Telephony Configuration Guide
10 Overview

Network Quality of Service

Enterprise Edge VoIP Gateway uses a method like the ITU-T Recommendation G.107, the E-Model, to determine the voice quality. This model evaluates the end­to-end network transmission performance and outputs a scalar rating “R” for the network transmission quality. The packet loss and latency of the end-to-end network determine “R”. The model correlates the network objective measure “R”, with the subjective QoS metric for voice quality, MOS or the Mean Opinion Score.
This model provides an effective traffic building mechanism by activating the Fallback to Circuit-Switche d Voice Fa ciliti es fea ture at call se t up to a void qualit y of service degradation. New calls fall back when the configured MOS values for all codecs are below the threshold.
The model is the reason for compression characteristics of the codecs. Each codec delivers a different MOS for the same network quality.

Network Monitoring

The VoIP Gateway network monitoring function measures the quality of service between the local and all remote gateways on a continuous basis. The network monitoring function exchanges UDP probe packets between all monitored gateways to collect the network statistics for each remote location. All the packets make a round trip from the Sender to Receiver and back to the Sender. From this information, you can calculate the latency and loss in the network for a distinct location.
Note 1: Quality of Service monitoring is not supported for non-Enterprise Edge product locations and must be disabled.
Note 2: The Quality of Service threshold is configurable per remote gateway. Note 3: Fallback starts for all new originating calls if the QoS of any monitored
gateway is below its threshold. Note 4: The fallback decision is made only at the originating gateway using the QoS
thresholds monitored at the originating gateway for the destination gateway. VoIP Gateway allows for manual configuration of QoS thresholds depending on the
customer preference between cost and voice quality. The Engineering guidelines chapter on page 15 provides the guidelines to determine the quality of service that supported for any given network.

Quality of service parameters

Quality of Service is depends on end-to-end network performance and available bandwidth. A number of parameters determine the VoIP Gateway QoS over the data network.
Enterprise Edge 2.0 IP Telephony Configuration Guide P0911590 Issue 02
Overview 11
The VoIP Gateway monitoring function can take about 3 mins to respond to marginal changes in the network condition. Fallback can be caused by any of the following reasons:
• Bad network conditions.
• The remote gateway is out of service.
• No network connection.
• Not enough DSP resources available.
Packet loss
Packet loss is the percentage of packets that do not arrive at their destination. Transmission equipment problems, and high delay and congestion can cause packet loss. In a voice conversation, gaps in the conversation represent packet losses. Some packet loss, less than 5%, can be acceptable without audible degradation in voice quality. Sporadic loss of small packets can be more acceptable than less frequent loss of large packets.
Packet delay
Packet delay is the time between when a packet leaves and when a packet arrives at it’s destination. The total packet delay time includes fixed and variable delay. Variable delay is the more manageable delay, while fixed delay depends on the network technology. The distinct network routing of packets are the cause of variable delays. The gateway must be as close as possible to the network backbone (WAN) with a minimum number of hops, to minimize packet delay and increase voice quality.
Delay variation (jitter)
The amount of variation in packet delay is otherwise known as delay variations, or jitter. Jitter affects the ability of the receiving gateway to assemble voice packets received at irregular intervals into a continuous voice stream.

Fallback to circuit-switched voice facilities

If the measured Mean Opinion Score (MOS) for all codecs is below the configured threshold for any monitored gateway, the Fallback to Conventional Circuit­switched services activates. This feature reroutes calls to different trunks such as the Public Switched Telephone Network (PSTN) until the network QoS improves. When the QoS meets or exceeds the threshold, calls route over the IP network.
Disable the fallback feature in the Local Gateway Configuration. With the fallback feature disabled, calls move across the IP telephony trunks no matter the QoS. The fallback feature is only active at call setup. A call in progress does not fall back if the QoS degrades.
Calls fallback if there is no response from the destination, an incorrectly configured remote gateway table, or if there are not enough DSP resources available to handle the new call.
P0911590 Issue 02 Enterprise Edge 2.0 IP Telephony Configuration Guide
12 Overview

Network Performance Utilities

There are two common network utilities, Ping and Traceroute. These utilities provide a method to measure quality of service paramet ers. Other utilities used also find more information about VoIP Gateway network performance.
Note 1: Because data network conditions can vary at different times, collect performance data over at least a 24 hour time period.
Note 2: Use performance utilities to measure performance from each gateway to every other gateway.
Ping
Ping (Packet InterNet Groper) sends an ICMP (Internet Control Message Protocol)
echo request message to a host, expecting an ICMP echo reply which allows for the measurement of a round trip time to a selected host. By sending repeated ICMP echo request messages, percent packet loss for a route can be measured.
Traceroute
Traceroute uses the IP TTL (time-to-live) field to determine router hops to a
specific IP address. A router must not forward an IP packet with a TTL field of 0 or
1. Instead, a router discards the packet and returns to the originating IP address an
ICMP “time exceeded” message.
Traceroute uses this mechanism by sending an IP datagram with a TTL of 1 to the
selected destination host. The first router to handle the da tagram sends back a “ time exceeded” message. This message identifies the first router on the route. The
Traceroute transmits a datagram with a TTL of 2.
Following, the second router on the route returns a “time exceeded” message until all hops are identified. The Traceroute IP datagram has a UDP Port number not likely to be in use at the destination (normally > 30,000). The destination returns a “port unreachable” ICMP packet. The destination host is identified.
Traceroute is used to measure round trip times to all hops along a route,
identifying bottlenecks in the network.

Codecs

The term codec refers to the voice coding and compression algorithm used by the DSP on the telephony services and the MSPECs. See the Enterprise Edge Programming Operations Guide for additional information on DSP and MSPEC resources.
The codec type for a VoIP Gateway call basis is determined at call setup. The originating gateway indicates to the remote gateway which codec types it supports, starting with the selected order of use. The remote gateway, depending on its capabilities, selects one of the codec types and continues with the call. If both ends cannot agree on a codec type, the call fails.
Enterprise Edge 2.0 IP Telephony Configuration Guide P0911590 Issue 02
Overview 13
All gateways in the intranet must use the same codec types. Each gateway is configured with available codecs with the selected order of use.
The codecs configuration must reflect available bandwidth on the network. Codec options are between quality compared to bandwidth.
The supported codec types are configured in the Modifying the Local Gateway
Configuration table section on page 65. The G.711 codec provides the best audio
quality but uses the greatest amount of bandwidth. The G.729 and G.723.1 codecs use less bandwidth, but reduce audio quality. The installer or administrator determines the best option for the user and the available bandwidth on the intranet. For example, if the WAN link cannot support multiple 64 kbit/s calls, G.711 must not be configured as a supported codec.
Enterprise Edge supports and recommends the following order for codec selection:
• G.729
• G.723.1 (6.3 kbit/s or 5.3 kbit/s)
• G.711
The G.729 codec provides the best balance of quality audio plus bandwidth savings. For more information about codecs, see the Codec types section on page 37.

Silence compression

G.723.1 and G.729, Annex B support Silence compression. A key to VoIP Gateway’s success in business applications is reducing WAN
bandwidth use. Beyond speech compression, the best bandwidth reducing technology is silence compression, also known as silence suppression. Silence compression technology identifies the periods of silence in a conversation, and stops sending IP speech packets during those periods. Telco studies show that in a typical telephone conversation, only about 36-40% of a full-duplex conversation is active. When one person talks, the other listens (known as half-duplex). And there are important periods of silence during speaker pauses between words and phrases.
For more information about silence compression, see the Silence compression section on page 38.

Echo cancellation

When a two-wire telephone cable connects to a four-wire PBX interface or a central office (CO) interface, the system uses hybrid circuits to convert between two wires and four wires. Although hybrid circuits are very efficient in their conversion ability, a small percentage of telephony energy is not converted but instead is reflected back to the caller. This is called echo.
For more information about echo cancellation, see the Echo cancellation section on page 40.
P0911590 Issue 02 Enterprise Edge 2.0 IP Telephony Configuration Guide
14 Overview

Jitter buffer

A major cause to reduced voice quality is IP network packet delay and network jitter. Network delay represents the average length of time for a packet to move across a network. Network jitter represents the differences in arrival time of a packet. Both important in de termining voice quality, de lay is like the av erage, jitter is like the standard deviation.
For more information about jitter buffer, see the Jitter buffer section on page 40.

Fax calls

The Enterprise Edge gateways support T.30 Group 3 fax calls. Fax calls automatically use the G.711 codec and require the associated bandwidth.
For more information about fax calls, see the Fax calls section on page 41.

Alarm Notification

Enterprise Edge uses the Unified Manager to record information about its working status.
See the Maintenance chapter on page 69 for additional information.
Enterprise Edge 2.0 IP Telephony Configuration Guide P0911590 Issue 02

Engineering guidelines

The engineering guidelines address the design of an IP trunk network for Enterprise Edge VoIP Gateway. The network contains the following:
• Enterprise Edge VoIP gateways
• Gateways attached to LANs
• Corporate intranet connecting the LANs
The guidelines assume that an installed corporate intranet connects the sites of the IP gateways.

Introduction

IP telephony compresses PCM voice and routes the packetized data over an intranet, to provide virtual analog TIE trunks between gateways. As voice traffic flows through at low marginal cost over existing private IP network facilities with available bandwidth on the private Wide Area Network (WAN) backbone, communication costs are lower.
This chapter provides guidelines for correctly designing a network of IP gateways over the corporate intranet. The chapter describes how to qualify the corporate intranet to support an IP network. This chapter indicates which changes ensure the maintenance of the quality of voice services when transferring those services from the PSTN. This chapter also addresses requirements for the successful integration with a customer's existing local area network (LAN). By following these guidelines the installer can configure the IP to ensure the best cost and quality and within a calculated tolerance.

Enterprise Edge IP telephony

Enterprise Edge IP telephony functions on a correctly provisioned and stable LAN. Delay, delay variation or jitter, and packet loss must be minimized end-to-end across the LAN and WAN. The installer must determine the design and configuration of the LAN and WAN that link the IP telephony system. If the intranet exceeds it’s capacity, new calls to the IP telephony fall back to conventional circuit-switched voice facilities to ensure the quality of service for new calls.

Overview

Traditional networks depend on voice services such as LEC and IXC private lines. With Enterprise Edge IP telephony technology, IP telephony selects a new kind of delivery mechanism that uses packet switching over a data network, a corporate intranet. The IP gateway converts a steady-stream digital voice into fixed length IP packets.
16 Engineering guidelines
Correct design procedures and rules are a must if a corporate network is expected to deliver voice traffic. The intranet introduces limits, delay, delay variation, and error, at levels that are higher than those delivered by voice networks. Delay between a user talking and a listener reduces the performance of conversations, while delay variation and packet errors introduce glitches in conversation. The connection of the IP gateways to the corporate intranet without preliminary evaluation can result in not acceptable degradation in the voice service.
A good design of the network begins with an understanding of traffic, and the network that carries the traffic. There are t hree preliminary steps whe re the installer must begin:
• Determine bandwidth requirements. The installer must determine the amount of traffic that the Enterprise Edge product will route through the IP gateway. This in turn places a traffic load on the corporate intranet. To determine bandwidth requirements, refer to the Enterprise Edge VoIP Gateway bandwidth
engineering section on page 17.
• Determine WAN link resources. If there are not enough resources in the corporate intranet to support voice services, the problem is normally because of not enough WAN resources. To determine WAN resources, refer to the
Determining WAN link resources section on page 23.
• Measure the existing in tranet’s QoS. The installer must determine the qual ity of voice service the corporate intranet can deliver. The Measuring Intranet QoS section on page 27 describes how to measure the delay and error characteristics of an intranet.
After the examination phase, the installer designs and installs the IP telephony network. This design not only includes the IP telephony but can also include making design changes to the intranet.
• The Further network analysis section on page 31 provide guidelines for modifying the intranet.
• The Implementing the network section on page 36 provides guidelines for integrating the IP gateway into the corporate LAN.
Engineering guidelines 17
Figure 1 IP Telephony network engineering process shows when the design and planning decisions that must occur.
Figure 1 IP Telephony network engineering process
Start
Determine bandwidth requirements
Assess WAN resources
Implement IP telephony
Network monitoring and data collection
Within QoS objectives?
No
Yes
Yes
Capacity available?
Yes
Measure intranet QoS
Within Qos expectations?
No
No
Further network analysis or design
Implement network changes

Enterprise Edge VoIP Gateway bandwidth engineering

Traffic controls the network design and the design process starts with the process of getting an IP telephony bandwidth forecast. The bandwidth forecast drives the following:
• LAN requirements (LAN must be great enough for the number of calls plus the overhead)
• WAN requirements (WAN must be great enough for the number of calls plus the overhead)
18 Engineering guidelines
Table 1 LAN and WAN IP bandwidth usage per Enterprise Edge Gateway (loaded
to 36 CCS per port per hour) with silence compression on page 20 and Table 2 LAN and WAN IP bandwidth usage per Enterprise Edge Gateway (loaded to 36 CCS per port per hour) without silence compression on page 21 show the bandwidth use for
the different codecs. This data assumes t hat each port is complete to 36 CCS (Centi­call-second). CCS is a channel or circuit occupied for 100 s. The worst case scenario is 100% utilization, or 36 CCS. Engineering the network for worst case numbers ensures that the network can handle peak traffic.

Multiple network interfaces

The Enterprise Edge can have more than one IP address. Figure 2 Multiple network
interfaces has three Enterprise Edge systems, each with more than one IP address
available. Define the IP address for the VoIP gateway in the Local Gateway table. The other remote gateways use this address to communicate with the Enterprise Edge VoIP Gateway.
Figure 2 Multiple network interfaces
Enterprise Edge 1
Local Gateway Table
Local Gateway IP IP2
Remote Gateway Table Remote Gateway IP addresses IP3 IP5
DN-A DN-B
IP2
IP1
DN-C
LAN/WAN
IP6IP7
IP5
Enterprise Edge 2
Local Gateway Table
Local Gateway IP IP3
Remote Gateway Table Remote Gateway IP addresses IP2 IP5
IP3
IP4
Enterprise Edge 3
Local Gateway Table
Local Gateway IP IP5
Remote Gateway Table Remote Gateway IP addresses IP3 IP3
Engineering guidelines 19
When a user at DN-A calls DN-C, the IP addresses in the Local Gateway and Remote Gateway tables of the each Enterprise Edge systems determine the call routing. The Remote Gateway table of the calling party (EE 1) contains the IP address where the outgoing voice packets are sent (EE 3). The Local Gateway table of EE 1 contains the IP address where EE 3 sends the return voice packets. The Local Gateway table of EE 3 contains the IP address which receives the voice packets from EE 1. The Remote Gateway table of EE 3 contains the IP address where it sends the return voice packets.
There are two methods to set up an IP address.

Method 1

On a routable internal LAN, assign the LAN IP address as the IP address in the Local Gateway table. See the Configuration chapter on page 61 for additional information on entering the Local Gateway IP address.

Method 2

In cases where the LAN is not routable, specify a WAN IP address. If you assign a WAN link as the local gateway IP address and the primary link fails, the VoIP function is lost. See the Configuration chapter on page 61 for additional information on entering the Local Gateway IP address.
For more information, see also the Enterprise Edge Programming Operations Guide.
20 Engineering guidelines

LAN engineering

Engineering the network for worst case numbers indicates the spare bandwidth a LAN must have to handle peak traffic. It is important the LAN be planned to handle the IP telephony traffic using the defined codec, without Ethernet delay or packet loss. The installer or administrator must select one configuration and then set up the LAN so there is more bandwidth than the IP telephony output.
Refer to standard Ethernet engineering tables for passive 10BaseT repeater hubs. Refer to the manufacturer’s specification for intelligent 10BaseT layer switches.
Table 1 LAN and WAN IP bandwidth usage per Enterprise Edge Gateway (loaded to 36 CCS per port per hour) with silence compression
Codec Type Packet
duration in ms (payload)
6
G.729 (8 kbit/s)
G.723.1 (5.3 kbit/s)
G723.1 (6.3 kbit/s)
Note 1: Note 2: Note 3: Note 4:
Ethernet frame over IP packet is 26 bytes.
Note 5:
frame. This gap is not included in the above bandwidth calculation.
Note 6: Note 7:
ured on the M1-ITG).
10 10 50 76 60.8 20.0 20 20 60 86 34.4 12.0 30 30 70 96 25.6 9.3 30 20 60 86 22.9 8.0
30 24 64 90 24.0 8.5
LAN data rate is the effective Ethernet bandwidth use. LAN kbit/s = Ethernet frame bytes*8*1000/Frame duration in ms 50% voice traffic reduction due to silence compression; no compression for fax. Overhead of (RTP+UDP+IP) packet over voice packet is 40 bytes; overhead of
Keep Ethernet bandwidth to support an Interframe gap of at least 12 bytes per
IP telephony uses a frame duration of 20 ms for G.729. If interworking with an M1-ITG, other frame durations are supported (config-
Voice/fax payload in bytes
IP packet in bytes
Ethernet
4
frame bytes
4
Bandwidth usage on LAN in kbit/s
Bandwidth usage on WAN in kbit/s
7
7

Silence compression

If an IP gateway acts as a tandem switch in a network where circuit-switched trunk facilities have a large amount of low audio level, enabling silence compression (also known as Voice Activity Detection) degrades the quality of service, causing broken speech. Under tandem switching conditions, with a large amount of low audio level, disable the silence compression using the IP telephony interface.
With silence compression disabled, the bandwidth use of the LAN/WAN approximately multiplies by two. Table 2 LAN and WAN IP bandwidth usage per
Enterprise Edge Gateway (loaded to 36 CCS per port per hour) without silence compression on page 21 shows the full-duplex bandwidth requirements with
silence compression disabled. Fax calls use a G.711 codec which does not support silence compression. Fax calls
require 64 kbit/s bandwidth.
Engineering guidelines 21
For more information about silence compression and fax calls, see the Silence
compression on page 38 and the Fax calls section on page 41.
Table 2 LAN and WAN IP bandwidth usage per Enterprise Edge Gateway (loaded to 36 CCS per port per hour) without silence compression
Codec Type Packet
duration in ms (payload)
6
G.711 (64 kbit/s)
G.729 (8 kbit/s)
G.723.1 (5.3 kbit/s)
G723.1 (6.3 kbit/s)
10 80 240 292 233.6 96 20 160 400 452 180.8 80 30 240 560 612 163.2 74.6
6
Note 1: Note 2: Note 3: Note 4:
Ethernet frame over IP packet is 26 bytes.
Note 5:
bytes per frame. This gap is not included in the above bandwidth calcu lation.
Note 6: Note 7:
ured on the M1-ITG).
10 10 100 152 121.6 40.0 20 20 120 172 68.8 24.0 30 30 140 192 51.2 18.6 30 20 120 172 45.8 16.0
30 24 128 180 48.0 17.0
LAN data rate is the effective Ethernet bandwidth use. LAN kbit/s = Ethernet frame bytes*8*1000/Frame duration in ms 50% voice traffic reduction due to silence compression; no compression for fax. Overhead of (RTP+UDP+IP) packet over voice packet is 40 bytes; overhead of
Keep Ethernet bandwidth to support to support an Interframe gap of at least 12
IP telephony uses a frame duration of 20 ms for G.729 and G.711. If interworking with an M1-ITG, other frame durations are supported (config-
Voice/fax payload in bytes
IP packet in bytes
Ethernet
4
frame bytes
Bandwidth
4
usage on LAN in kbit/s
Bandwidth usage on WAN in kbit/s
7
7 7
7
Example 1: LAN engineering  voice calls
Consider a site with four Enterprise Edge IP telephony ports. The Preferred codec is G.729, using a voice payload of 20 ms. Silence compression is enabled.
Given the above, what is the peak traffic in kbit/s that IP telephony will put on the LAN?
With Table 1 LAN and WAN IP bandwidth usage per Enterprise Edge Gateway
(loaded to 36 CCS per port per hour) with silence compression on page 20, for calls
with silence compression, each port generates 34.4 kbit/s when engaged in a call to another gateway. If all four ports are in use, then the additional load is 137.6 kbit/s.
Example 2: LAN engineering  fax calls
Consider a site with four IP telephony ports. The required codec is G.711, with a voice payload of 20 ms. Silence compression is not used.
22 Engineering guidelines
With Table 2 LAN and WAN IP bandwidth usage per Enterprise Edge Gateway
(loaded to 36 CCS per port per hour) without silence compression on page 21, for
calls without silence compression, each fax call generates 180.8 kbit/s. If all four ports are in use for fax calls, then the additional load is 723.2 kbit/s. For more information about fax calls, see the Fax calls section on page 41.

WAN engineering

To get traffic to Wide Area Network (WAN), use the formula: 0.5 x IP packet in bytes x 8 x 1000/payload in ms. The reason for the reduction data rate is because of the type of a duplex channel on a WAN. For example, with G.711 codec, a two-way conversation channel has a rate of 128 kbit/s. However, the same conversation on WAN (for example, a T1) requires a 64 kbit/s channel only, because a WAN channel is a full duplex channel.
Both “talk” and “listen” traffic use a part of the 10 Mbit/s Et hernet channel while a conversation uses a 64 kbit/s (DS0) duplex channel in a T1 or other WAN media.
Example 1: WAN engineering  voice calls
Consider a site with four IP telephony ports. The Preferred codec is G.723.1,
6.3 kbit/s, using a voice payload of 30 ms. Silence compression is enabled.
Given the above, what is the peak traffic in kbit/s that IP telephony will put on the WAN?
With Table 1 LAN and WAN IP bandwidth usage per Enterprise Edge Gateway
(loaded to 36 CCS per port per hour) with silence compression on page 20, for
silence compression, each port generates 8.5 kbit/s when engaged in a call. If all four ports are in use, then the additional load is 34 kbit/s.
Example 2: WAN engineering  fax calls
Consider a site with four IP telephony ports. The G.711 codec automatically selected, with a voice payload of 20 ms. Silence compression is not used in the G.711 codec.
With Table 2 LAN and WAN IP bandwidth usage per Enterprise Edge Gateway
(loaded to 36 CCS per port per hour) without silence compression on page 21, for
no silence compression, each port generates 80 kbit/s when engaged in a call. If all four ports are in use, then the additional load is 320 kbit/s.
Engineering guidelines 23

Determining WAN link resources

For most installations, IP telephony traffic is routed over WAN links within the intranet. WAN links are the most expensive recurring expenses in the network and they often are the source of capacity problems in the network. Different from LAN bandwidth, which is almost free and easily installed, WAN links, especially inter­LATA and international links require time to receive financial approval, provision and upgrade. For these reasons, it is important to determine the state of WAN links in the intranet before installing the IP telephony.
Each voice conversation, (G.729, Annex B codec, 20 ms payload) uses 12 kbit/s of bandwidth for each link that moves across in the intranet; a DS0 can support below 5 simultaneous telephone conversations.

Link utilization

The starting point of this evaluation is to get a current topology map and link utilization report of the intranet. A visual inspection of the topology map indicates which WAN links are expected to deliver IP telephony traffic. Also use the
Traceroute tool (see Measuring Intranet QoS on page 27).
The next step is to find out the current utilization of those links. Note the reporting window that appears in the link utilization report. For example, the link utilization can be an average of a week, a day, or one hour. To be consistent with the dimensioning considerations (see Enterprise Edge VoIP Gateway bandwidth
engineering on page 17), get the peak utilization of the trunk. Also, because WAN
links are full-duplex that data services show asymmetric traffic behavior, get the utilization of the link representing traffic flowing in the heavier direction.
The third step is to determine the available spare capacity. Enterprise Edge intranets are subject to capacity planning controls that ensure that capacity use remains below some determined utilization level . For ex ampl e a pla nning control can sta te that the utilization of a 56 kbit/s link during the peak hour must not exceed 50%; for a T1 link, the threshold is higher, for example at 85%. The carrying capacity of the 56 kbit/s link can be 28 kbit/s, and for the T1 1.3056 Mbit/s. In some organizations the thresholds can be lower than that used in this example; in the event of link failures, there needs to be spare capacity for traffic to be re-routed.
Some WAN links can be provisioned on top of layer 2 services such as Frame Relay and ATM; the router-to-router link is a virtual circuit, which is subject not only to a physical capacity, but also a “logical capacity” limit. The installer or adm inistrator needs to get the physical link capacity and the QoS parameters. The important QoS parameters are CIR (committed information rate) for Frame Relay, and MCR (maximum cell rate) for ATM.
The difference between the current capaci ty and its acceptable l imit is the availa ble capacity. For example a T1 link used at 48% during the peak hour, with a planning limit of 85% has an available capacity of about 568 kbit/s.
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