Avaya EE IP Telephony Configuration manual

Enterprise Edge IP Telephony Configuration Manual

PO908509 Issue 01

Chapter 1 Overview 7

About this document 8 System Functions 8 Dialing Plan Support 9

Enterprise Edge IP telephony and M1 networking 9 Toll bypass with VoIP Gateway 12

Network Quality of Service 16

Network Monitoring 17 Quality of service parameters 17

Fallback to circuit-switched voice facilities 18 Network Performance Utilities 18 Codecs 19 Silence compression 22 Echo cancellation 23

Non-linear processing 24 Jitter buffer 24 Fax calls 25 Alarm Notification 26

Contents 3

Chapter 2 Engineering guidelines 27

Introduction 27

Enterprise Edge IP telephony 27 Overview 28 Enterprise Edge VoIP Gateway bandwidth engineering 29 Multiple network interfaces 30

Method 1 31

Method 2 32 LAN engineering 32

Silence compression 33 WAN engineering 35

Assessing WAN link resources 35

Link utilization 36

Estimating network loading due to IP telephony traffic 36

Other intranet resource considerations 38 Setting QoS 39 Measuring Intranet QoS 40

Measuring end-to-end network delay 40

Measuring end-to-end packet loss 41

Recording routes 41

Adjusting ping measurements 42

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Measurement procedure 43

Other measurement considerations 43 Further network analysis 44

Components of delay 44

Reducing link delay 45

Reducing hop count 46

Routing issues 47 Implementing QoS in IP networks 47

Traffic mix 48

TCP traffic behavior 49

Enterprise Edge Router QoS Support 49 Implementing the network 49

LAN engineering 49

Fallback threshold 52 Post-installation network measurements 53

Setting IP telephony QoS objectives 53

Intranet QoS monitoring 54

User feedback 54 Dialing plan 55

IP telephony and M1 networking 55

Toll bypass with IP telephony 58

Core telephony services configuration 62

Chapter 3 Engineering checklist 65

Chapter 4 Installation 67

Installation Roadmap 67

Configuring the local gateway 67

Adding a remote gateway 68

Chapter 5 Configuration 73

User Interface Overview 74

Local gateway configuration 75

Remote gateway configuration 78

Core telephony services configuration 79

Configuration of fallback to conventional circuit-switched facilities 80

Chapter 6 Maintenance 81

Quality of Service Monitor 81

Quality of Service Status 81

Using the QoS Monitor pull-down View menu 81 Operational Statistics 81 Backup and Restore Procedures 81

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Chapter 7 Interoperability 83

interoperability considerations 83

Asymmetrical media channel negotiation 84

No feedback busy station 84

Glossary 85

Index 87

Contents 5

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6 Contents

Enteprise Edge Unified Messaging Client Installation Guide P0908532 Issue 01

Overview

The Enterprise Edge VoIP Gateway reduces customers’ 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 integral part of the telephony services. IP telephony is transparent to users.

Enterprise Edge provides IP telephony capability. IP telephony involves the conversion of voice from its traditional telephony format (continuous analog or digital signal) into a digital packet format that can be transported over an intranet.

IP telephony operates on an installed corporate IP network. IP telephony requires a well managed intranet, rather than the internet. The private IP network facilities must have under-utilized bandwidth on the private Wide Area Network (WAN) backbone. The Engineering guidelines chapter of this guide contains information on 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 of this guide for information on how to configure IP telephony.

VoIP Gateway supports ITU-H.323v2 gateway operation. VoIP Gateway uses standard Digital Signal Processor (DSP) voice coding. See the Enterprise Edge Programming Operations Guide for information on DSP. VoIP Gateway supports compression algorithms (codecs) such as G.711, G.723, and G.729. See Codec types in the Engineering guidelines chapter for information on codecs.

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 declines. 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 and Configuration of fallback to conventional circuit-switched facilities sections in the Configuration chapter for information on configuring the QoS parameters. If the quality falls below the expected level of QoS, the regular c ircuit-switched voi ce facilitie s route is selecte d until the QoS returns to an acceptable level.

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About this document

This guide provides information on the Enterprise Edge VoIP Gateway. This guide is addressed to both telecom and datacom engineers who are going to design and implement the network. It is assumed that the telecom engineer is familiar with engineering the Enterprise Edge product portfolio, and obtaining system voice and fax traffic statistics. It is assumed that the datacom engineer is familiar with 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 the person in either the telecom or datacom engineering role. This guide contains the following sections:

• Engineering guidelines

• Engineering checklist

• Installation

• Configuration

• Maintenance

• Interoperability

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.323 v2 compatible gateway

• 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 offered through Enterprise Edge treats 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 integral part of the Enterprise Edge product portfolio.

VoIP Gateway trunks are supervised trunks with answer and disconnect supervision. The VoIP Gateway supports voice and fax calls. See the Engineering guidelines chapter for more information about fax calls. VoIP Gateway does not support modem calls.

Enterprise Edge IP Telephony Configuration Manual PO908509 Issue 01

The IP telephony gateway allows communication with other supported gateways and H.323 v2 gateways through trunk calls. The IP telephony gateway supports Direct Routed communication. The local gateway performs the address resolution. The local gateway maintains the remote gateway table.

Dialing Plan Suppor t

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 decide which trunking facilities are used for calls and when they are used.

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.

Overview 9

See the Configuration chapter for details on configuring destination digits and destination codes.

The dialing plans for all VoIP Gateways connected to the corporate intranet need to be coordinated so that calls can be made between gateways as required.

Enterprise Edge IP telephony and M1 networking

This example shows a private network composed of one central Meridian 1, and two smaller sites with Enterprise Edge systems connected over IP trunks through a corporate IP network. This could represent a large head office (with the Meridian

1) connected to several smaller branch offices. In this network, only the head office has trunks connected to the public network.

The branch offices access the public network using IP trunks to the head office. This configuration allows for cost savings by consolidating the public access trunks. Users at all three locations access the public network by dialing ‘9’, followed by the public number. For example, a user in the west end branch might dial 9-555-1212 (for a local call) or 9-1-613-555-1212 (for a long distance call). These public calls are routed to the Meridian 1 by the Enterprise Edge routing table. Routing tables at the Meridian 1 will then select an appropriate public facility for the call.

Private network calls are made by dialing a 4-digit private network DN. For example, if a user in the west end branch wishes to call a user in the east end branch within the private network, they dial 6221.

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Figure 1 Enterprise Edge and M1 networking overview

Network # 2221 Received # 2221 Internal # 2221

Enterprise Edge West end Branch I.P. Address 192.1.1.2

Call Managers: West end: 6 192.1.1.3

4 192.1.1.4 East end: 2 192.1.1.2

4 192.1.1.4

WAN

IP IP

Meridian M1 DN: 4221

PRI (public protocol)

Central Office

I.P. Address 192.1.1.4

Network # 6221 Received # 6221 Internal # 6221

Enterprise Edge East end Branch I.P. Address 192.1.1.3

Note: The quality of the IP trunk connection is assessed during initial call setup, and if the quality is poor, Enterprise Edge will try to find an alternate route to complete the call (fallback) based on the programming definitions in the routing table. For simplicity, this example does not show programming for fallback. In this example, if the quality of the IP connection is considered too low during the call setup phase, the call would fail. For an example of fallback programming, refer to the section, “Toll bypass with VoIP Gateway” on page 12.

Note: Enterprise Edge VoIP Gateway requires a keycode. After entering the keycode for Enterprise Edge IP Telephony, perform a warm reset by following the procedure in the Maintenance section of the Enterprise Edge Programming Operations Guide.

In the table that follows, private network routing information is highlighted in gray. Public network routing information is shown in white.

The gateways examine the dialed digits and route the call to the corresponding IP address.

Heading Parameter Setting

West End office: Trunk/Line Data Line 241 Target line

Received # 2221

Line Access Set 2221 L241:Ring only

Line pool access Line pool A

To Head office (M1):

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Heading Parameter Setting

Service/Routing Service Route 001

Use Pool A External # (leave blank) DN type Private Destination Code 4 Normal route 001

Absorb None To East End: Service/Routing Service Destination Code 6

Normal route 001

Absorb None To Public Network: Service/Routing Service Route 002

Use Pool A

External # (leave blank)

DN type Public

Destination Code 9

Normal route 002

Absorb None

Overview 11

East End office: Trunk/Line Data Line 241 Target line

Received # 6221 Line Access Set 6221 L241:Ring only

Line pool access Line pool A To Head Office: (M1) Service/Routing Service Route 001

Use Pool A

External # (leave blank)

DN type Private

Destination Code 4

Normal route 001

Absorb None To West End: Service/Routing Service Destination Code 2

Normal route 001

Absorb None To Public Network: Service/Routing Service Route 002

Use Pool A

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Heading Parameter Setting

In this example, outgoing public network calls dialed from a Enterprise Edge Voice Soution set are passed to the Meridian M1, and the Meridian M1 i s responsibl e for seizing a public trunk. For this reason, the ‘9’ prefix is left in the number passed to the Meridian 1.

Note: Ensure that Line Pool A is used for IP trunks. In order for the digit counting algorithm for outgoing IP calls to take into account

this extra digit, the Private Network Access Code must be set to ’9’ on each Enterprise Edge system.

The Meridian M1 must recognize incoming 2xxx and 6xxx DID calls, and route the call over IP trunks to either the East or West end offices.

External # (leave blank)

DN type Public

Destination Code 9

Normal route 002

Absorb None

The Meridian M1 must recognize numbers starting with ‘9’ as public numbers, whether the numbers are dialed by Meridian M1 users or by Enterprise Edge Voice users.

Toll bypass with VoIP Gateway

This example shows a private network composed of one Enterprise Edge in Toronto and one Enterprise Edge in Ottawa, connected over IP trunks through a corporate IP network.

In this network, each Enterprise Edge has a PRI trunk to the Central Office, and IP trunks to the other Enterprise Edge. Calls from the Toronto system to the Ottawa system and the Ottawa public network are made over IP trunks with fallback to the PRI trunks when IP trunks are congested. This configuration allows for cost savings by using the corporate IP network whenever possible, thereby bypassing toll charges that would be incurred by using the public network.

Note: When a call gets rerouted over the PSTN due to congestion, the user may see a prompt "Expensive route." The warning indicates that toll charges may be applied to this call.

Users at both locations access the public network by dialing ’9’, followed by the public number. For example, a user in Toronto might dial 9-555-1212 (for a local call), or 9-1-613-555-1212 (for a long distance call to Ottawa). Local calls would be sent directly to the Central Office over PRI trunks. Long distance calls to Ottawa would be sent over IP trunks; the Ottawa system would tandem these calls to the local Central Office over PRI trunks.

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Overview 13

Private network calls are made by dialing a 4-digit private network DN. For example, if a user in Toronto wants to call a user in Ottawa within the private network, they dial 6221.

Figure 2 Toll bypass overview

PRI (public protocol)

Public Network

Network # 6221 Received # 6221 Internal # 6221

Network # 2221 Received # 2221 Internal # 2221

PRI (public protocol)

Public Network

Corporate I.P. Network

IP IP

Enterprise Edge Toronto Branch I.P. Address 192.1.1.2

Enterprise Edge Ottawa Branch I.P. Address 192.1.1.3

The Gateway at the Toronto office examines the dialed digits and determines that it should be routed to the IP address corresponding to the Ottawa office. The Ottawa office receives the call, sees that the leading digit(s) match its Private Network Access Code, and uses a destination code to route the call over its public trunks to the PSTN.

This is a simplified example where only calls to the 613 Area Code are routed by the Ottawa node. In a real world configuration, it would also be desirable to handle Area Codes that are ‘close’, for example Montreal: 514.

In the table that follows, private network routing information is highlighted in gray. Public network routing information is shown in white.

Heading Parameter Setting

Toronto office: Lines/Trunk/Line Data Line 241 Target line

Received # 2221 Terminals & sets/Line Access Set 221 L241:Ring only

Line pool access Line pool A

Line pool PRI-A

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Heading Parameter Setting

Calls to Ottawa office: Services/Routing Service Route 001

Services/Routing Service Route 002

Services/Routing Service Destination Code 6

Calls to Ottawa Public Network: Services/Routing Service Route 003

To Public Network: Services/Routing Service Destination Code 9161A

Use Pool A

External # (leave blank)

DN type Private

Use Pool PRI-A

External # (leave blank)

DN type Private

Schedule 4 001

Absorb None

Normal route 002

Absorb None

Use Pool A

External # (leave blank)

DN type Public

Route 004

Use Pool PRI-A

External # (leave blank)

DN type Public

Destination Code 91613

Normal route 004

Absorb 1

Schedule 4 003

Absorb None

Normal route 004

Absorb 1

Destination Code 916A

Normal route 004

Absorb 1

Destination Code 91A

Normal route 004

Absorb 1

Destination Code 9A

Normal route 004

Absorb 1

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Overview 15

Heading Parameter Setting

Ottawa office: Trunk/Line Data Line 241 Target line

Received # 6221 Line Access Set 6221 L241:Ring only

Line pool access Line pool A

Line pool PRI-A To Toronto office: Services/Routing Service Route 001

Use Pool A External # (leave blank) DN type Private Route 002 Use Pool PRI-A External # (leave blank) DN type Private Destination Code 2 Normal route 002 Absorb None Schedule 4 001

Absorb None To Toronto Public Network: Services/Routing Service Route 003

Use Pool A

External # (leave blank)

DN type Public Services/Routing Service Route 004

Use Pool PRI-A

External # (leave blank)

DN type Public

Destination Code 91416

Normal route 004

Absorb 1

Schedule 4 003

Absorb None To Public Network: Services/Routing Service Destination Code 9141A

Normal route 004

Absorb 1

Destination Code 914A

Normal route 004

Absorb 1

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Heading Parameter Setting

The implications on the configuration on each node are:

• each node must have the Private Network Access Code set to the value 9.

• each node must have destination code(s) that match the Private Network Access

Destination Code 91A

Normal route 004

Absorb 1

Destination Code 9A

Normal route 004

Absorb 1

Code plus digits corresponding to calls terminating in the local PSTN. For example, if the Private Network Access Code is ‘9’, the node in Ottawa would require a destination code of ‘91613’. Similarly, Toronto would require the following destination code: 91416.

Note: Ensure that Line Pool A is used for IP trunks.

• To allow for fallback to PRI trunks when the IP trunks are congested, you must also program the following Routing service settings:

• Set the start and end times for Sched 4 to 1:00 so that IP calls can be made 24 hours a day.

• Program the Sched 4 Service setting to Auto and enable overflow routing by changing the Overflow setting to Y (Yes).

• A control set must be defined for all sets on the system that make calls over IP trunks. See the Enterprise Edge Programming Operations Guide for more information.

You must program Remote Packages so that the IP trunks in Pool A can access the lines in Pool PRI-A in a toll bypass scenario. In other words, you must give package 01 access to pool PRI-A and you must assign package 01 to all IP trunks. For more information, see the Enterprise Edge Programming Operations Guide.

Network Quality of Service

Enterprise Edge VoIP Gateway uses a method similar to 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 further correlates the network objective measure “R”, with the subjective QoS metric for voice quality, MOS or the Mean Opinion Score.

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Overview 17

This model serves as an effective traffic shaping mechanism by invoking the Fallback to Circuit-Switched Voice Facilit ies feature at call set up to avoid quality of service degradation. New calls fall back when the configurable MOS values for all codecs fall below the threshold.

The model accounts 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, the latency and loss in the network for a particular location are calculated.

It may take about 3 mins before the VoIP Gateway monitoring function reacts to marginal changes in the network condition. Fallback can be due to any of the following reasons:

• Bad network conditions.

• The remote gateway is out of service.

• No network connection.

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 is triggered for all new originating calls if the QoS of any

monitored gateway falls 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 trade-off between cost and voice quality. The Engineering guidelines chapter provides the necessary guidelines to effectively weigh the trade-off and determine the quality of service that can be supported for any given network.

Quality of service parameters

Quality of Service is largely dependent on end-to-end network performance and available bandwidth. A number of parameters determine the VoIP Gateway QoS over the data network.

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Packet loss

Packet loss is the percentage of packets that do not arrive at their destination. Packet loss is caused by transmission equipment problems, and high delay and congestion. In a voice conversation, packet loss is heard as gaps in the conversation. Some packet loss, less than 5%, may be acceptable without too much degradation in voice quality. Sporadic loss of small packets may be more acceptable than infrequent loss of large packets.

Packet delay

Packet delay is the time between when a packet is sent and when it is received. The total packet delay time consists of fixed and variable delay. Variable delay is the more manageable delay, since fixed delay is dependent on the network technology itself. Variable delay is caused by the particular network routing of packets. The gateway should be as close as possible to the network backbone (WAN) with a minimum number of hops, to minimize packet delay and maximize voice quality.

Delay variation (jitter)

The amount of variation in packet delay is referred to 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 falls below the configured threshold for any monitored gateway, the Fallback to Conventional Circuit-switched services is triggered. This feature reroutes calls to alternate trunks such as the Public Switched Telephone Network (PSTN). The feature reroutes calls until the network QoS improves. When the QoS meets or exceeds the threshold, calls route over the IP network.

The fallback feature can be disabled in the Local Gateway Configuration. If the fallback feature is disabled, calls are sent over the IP telephony trunks regardless of the QoS. The fallback feature is only in effect at call setup. A call in progress will not fall back if the QoS degrades.

Network Performance Utilities

Two common network utilities, Ping and Traceroute, are described below. These utilities provide a method to measure quality of service parameters. Other utilities can be used to find more information about VoIP Gateway network performance.

Note 1: Since data network conditions can vary at different times, collect performance data over at least a 24 hour time period.

Note 2: Performance utilities should be used to measure performance from each gateway to every other gateway.

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Overview 19

Ping

Ping (Packet InterNet Groper) sends an ICMP (Internet Control Message Protocol)

echo request message to a host, expecting an ICMP echo reply to be returned. This allows the round trip time to a particular host to be measured. By sending repeated ICMP echo request messages, percent packet loss for a route can also 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. It must instead throw away the packet and return 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

specified destination host. The first router to handle the datagram will send back a “time exceeded” message. This identifies the first router on the route. The

traceroute sends out a datagram with a TTL of 2.

This will cause the second router on the route to return a “time exceeded” m essage and so on until all hops have been identified. The traceroute IP datagram will have a UDP Port number unlikely to be in use at the destination (usually > 30,000). This will cause the destination to return a “port unreachable” ICMP packet. This identifies the destination host.

Traceroute can be used to measure round trip times to all hops along a route,

thereby 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 used on a per VoIP Gateway call basis is determined at call setup. The originating gateway will indicate to the remote gateway which codec types it supports, starting with the preferred order of usage. The remote gateway, depending on its capabilities, chooses one of the codec types and continues with the call. If both ends cannot agree on a codec type, the call fails.

Therefore, it is important that all gateways in the intranet use the same codec types. Each gateway needs to be configured with which possible codecs are available for

negotiation, as well as the preferred order of usage. Given that the trade-off is quality versus bandwidth, the codecs configuration should reflect available bandwidth on the network.

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The supported codec types are configured in the Local gateway configuration section. 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 choice 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 should not be configured as a supported codec.

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Enterprise Edge Solutions 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. Enterprise Edge VoIP Gateway supports the following codecs:

G.711

This codec delivers “toll quality” audio at 64 kbit/s. This codec is optimal for speech since it has the smallest delay, and is very resilient to channel errors. However, it consumes the largest bandwidth. North America uses G.711 µ-LAW and international markets use G.711 A-LAW.

G.729

The G.729 codec is the default and preferred codec for IP telephony. It provides near toll quality with a low delay. This codec uses compression to 8 kbit/s. Enterprise Edge VoIP Gateway supports G.729 with silence compression, per Annex B.

G.723.1

The G.723.1 codec uses the smallest amount of bandwidth. This codec uses the greatest compression, 5.3 kbit/s or 6.3 kbit/s.

The G.723.1 codec uses a different compression method than the G.729 codec. The G.723.1 method uses more DSP resources. Each MSPEC supports only one G.723.1 call. A G.711 call can run in the same MSPEC as a G.723.1 call. See the Enterprise Edge Programming Operations Guide for additional information.

If the G.723.1 codec is the only possible codec for a call, a trunk may not be available for the call if there are insufficient DSP resources available. All VoIP Gateway facilities will appear to be in use, even though there are DSP resources available for calls using other codec types.

Since most gateways support the G.711 codec, configure G.711 as a supported codec. The G.711 codec does not compress audio or fax. The G.711 codec supports two IP trunks on each MSPEC. See the Enterprise Edge Programming Operations Guide for additional information.

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Silence compression

Silence compression is supported on G.723.1 and G.729, Annex B. A key to VoIP Gateway’s success in business applications is minimizing WAN

bandwidth consumption. Beyond speech compression, the best bandwidth reducing technology is silence compression, also known as silence suppression. Silence compression technology recognizes the periods of silence in a conversation , and stops sending IP speech packets during those periods. Telco studies show that in a typical phone conversation, only about 36-40% of a full-duplex conversation is active. When one person talks, the other listens (this is called half-duplex). And there are significant periods of silence during speaker pauses between words and phrases.

By applying silence compression, full duplex bandwidth consumption is reduced by the same amount, freeing up bandwidth for other voice/fax or data communications. The following figure illustrates how silence compression allows two conversations to fit in the bandwidth otherwise used by one. This 50% bandwidth reduction develops over a 20-30 second period as the conversation switches from one direction to another.

To provide a more natural sound, comfort noise is added at the destination gateway during the silent periods to ca lls where silence compression is active. In some cases, silence compression may cause a perceived degradation in audio quality. Silence compression can be disabled. Disabling silence compression will increase bandwidth consumption.

If VoIP Gateway serves as a tandem switch in a network where some circuitswitched trunk facilities have an excessively low audio level, silence compression, if enabled, will degrade the quality of service by causing choppiness of speech. Under these conditions, silence compression should be disabled.

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Echo cancellation

When a two-wire telephone cable connects to a four-wire PBX interface or a telco central office (CO) interface, a special electrical circuit called a hybrid is used 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.

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If the caller is near the PBX or CO switch, the echo comes back so quickly it cannot be discerned. However, if the delay is more than about 10 ms, the caller can hear an echo. To prevent this, gateway vendors include special code in the DSPs that listens for the echo signal and subtracts it from the listener’s audio signal. Echo cancellation is especially important for gateway vendors because the IP network delay can easily be 40−50 ms, so the echo from the far-end hybrid would be quite pronounced at the near end. Far-end echo cancellation eliminates this.

Echo cancellation sometimes causes choppiness in conversation in a low audio conversation. Although echo cancellation can be disabled, it is not recommended.

Non-linear processing

Non-linear processing (NLP) is part of echo cancellation. It improves echo cancellation by further reducing residual echo. NLP mutes background noise during periods of far-end silence and prevents comfort noise from being generated. Some listeners find muted background noise annoying. NLP can be disabled to prevent this, but with the trade-off of increased perceived echo.

Jitter buffer

A major contributor to reduced voice quality is IP network packet delay and network jitter. Network delay describes the average length of time for a packet to traverse a network. Network jitter describes the variability in arrival time of a packet. Delay is like the average, jitter is like the standard deviation. Both are important in determining voice quality.

To allow for variable packet arrival time and still produce a steady out-going stream of speech, the far-end gateway does not play out the speech as soon as the first packet arrives. Instead, it holds it for a certai n time in part of its me mory called the jitter buffer, and then plays it out. The amount of this hold time is the measure of the jitter buffer, e.g., a 50 ms hold time implies a 50 ms jitter buffer.

As the network delay (total time, including codec processing time) exceeds about 200 ms, the two speakers will increasingly adopt a half-duplex communications mode, where one speaks, the other listens and pauses to make sure the speaker is done. If the pauses are ill timed, they end up “stepping” on each other’s speech. This is the problem that occurs when two people converse over a satellite telephony connection. The result is a reduction in perceived voice quality.

When a voice packet is inordinately delayed and does not arrive at the far-end in time to fit into the voice stream going out of the far-end gateway, it is discarded, and the previous packet is replayed. If this happens too often, or twice in a row, the listener will perceive reduced voice quality.

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Overview 25

The jitter buffer hold time adds to the overall dela y, so if the network ha s high jitter, the overall effect will be a long perceived delay in the voice stream. For example, a network might have a moderately average delay of 50 ms and a variabilit y of 5 ms. The network is said to have 5 ms of jitter, a low figure. The jitter buffer hold time is only 5 ms, so the effective network total delay will only be 55 ms, still moderate.

On the other hand, if a network has a low average delay of 15 ms, but 10% of the time the delay goes out to a long 100 ms, while 90% of t he time the delay is a brief 4 ms, the jitter buffer would have to be 100 ms and the total effective network delay would be 115 ms, a long delay. Network jitter can be more important than average delay in many VoIP Gateway applications.

VoIP Gateway voice calls use an adaptive jitter buffer that changes the hold time over the duration of the call. The installer or administrator configures the maximum hold time.

VoIP Gateway fax calls use a fixed jitter buffer that does not change the hold tim e over the duration of the call. Fax calls are more sensitive to packet loss. In situations of high jitter, increased delay (through the use of a deeper jitter buffer) is preferred. To accommodate this, VoIP Gateway provides a separate jitter buffer setting for fax calls.

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.

As the gateway does not know in advance that a call will carry a fax transmission, it first establishes a voice channel. The voice channel may use G.729 or G.723.1 audio compression. Upon detecting the answering fax machine’s CED tone, the terminating gateway performs the following operations:

• Initiates the procedure to revert the speech path to a G.711, 64 Kbit/s clear

• Disables the adaptive jitter buffer feature.

• Sets the hold time for the jitter buffer to the value specified in the Local

The answering fax machine must produce its CED tone within 15 s of connection. The terminating gateway turns off CED tone detection after 15 s to prevent false tone detection during a voice call.

This method imposes the following restrictions:

• Interoperability with other IP gateways. A terminating gateway must support

channel.

Gateway settings to improve late IP packet tolerance.

CED fax tone detection, and initiate the procedure as described in previous paragraphs. An originating gateway must support the H.323 Request Mode procedure, but does not need to detect fax tones. The originating gateway must additionally be capable of supporting the large G.711 packest used for fax transmission.

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26 Overview

• In order for the gateways to revert to a G.711 clear channel, the terminating fax machine must issue a CED tone upon answering the call. Manually initiated fax transmissions, where the user at the terminating end first talks with the originating user before setting the te rminating fa x to rece ive the docume nt, are not supported.

• Fax machines tolerate a maximum round trip delay of 1200 ms. Media processing in the the two gateways introduces a round trip delay of approximately 300 ms, in addition to the delay caused by the jitter buffer. If a 250 ms jitter buffer is used, IP latency should never exceed (1200 -(300+(2*250))) = 400 ms round trip delay, or approximately 200 ms one way.

Alarm Notification

Enterprise Edge uses the Unified Manager to capture information about its operational status.

See the Maintenance chapter for additional information.

Enterprise Edge IP Telephony Configuration Manual PO908509 Issue 01

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 a private internet, or intranet, to provide virtual analog TIE trunks between gateways. Communications costs may be reduced as voice traffic is routed at low marginal cost over existing private IP network facilities with available under-utilized bandwidth on the private Wide Area Network (WAN) backbone.

This document provides guiding principles for properly designing a network of IP gateways over the corporate intranet, describe how to qualify the corporate intranet to support an IP network, and decide what required changes are needed in order to preserve the quality of voice services as much as possible when migrating those services from the PSTN. It addresses requirements for the successful integration with the customer's existing local area network (LAN). By adhering to these guidelines the designer should be able to engineer the IP so that the cost and quality trade-off is at best imperceptible, and at worst within a calculated tolerance.

Enterprise Edge IP telephony

Enterprise Edge IP telephony is designed to work on an adequately provisioned, stable LAN. Delay, delay variation or jitter, and packet lo ss must be minimized endto-end across the LAN and WAN. The installer must carefully determine the design and configuration of the LAN and WAN that link the IP telephony system. If the intranet becomes congested, new calls to the IP telephony will fall back to traditional circuit-switched voice facilities so that the quality of service is not degraded for new calls.

IP telephony operates on an installed corporate IP network. IP telephony operates on a well managed intranet, rather than the internet.

PO908509 Issue 01 Enterprise Edge IP Telephony Configuration Manual

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