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Avaya IP Telephony Implementation Guide
Communication Manager 3
...
Configurations and recommendations are given for various Avaya Media Servers and
Gateways, as well as Avaya 4600 Series IP Telephones
...

The intent of this document is to provide training on IP telephony, and to give guidelines for
implementing Avaya solutions
...

This document covers the Avaya Communication Manager 2
...
1, and the Avaya 4600 Series IP
Telephone 1
...

External posting: www
...
com
...
Although the information is
believed to be accurate, it is provided without guarantee of complete accuracy and without warranty of
any kind
...
Avaya shall
not be liable for any adverse outcomes resulting from the application of this document; the user accepts
full responsibility
...
cisco
...
Although all reasonable efforts have been made to provide accurate information
regarding Cisco products and features, Avaya makes no claim of complete accuracy and shall not be
liable for adverse outcomes resulting from discrepancies
...

© 2005 Avaya Inc
...

Avaya and the Avaya Logo are trademarks of Avaya Inc
...
, a wholly owned subsidiary
of Avaya Inc
...
All trademarks identified by ® and
™ are registered trademarks or trademarks, respectively, of Avaya Inc
...


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Avaya IP Telephony Implementation Guide

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Foreword
Several benefits are motivating companies to transmit voice communications over packet networks
originally designed for data
...
By eliminating a separate circuit-switched voice network, businesses avoid the
expenses of buying, maintaining and administering two networks
...

Another benefit is the potential to more tightly integrate data and voice applications
...
A converged multi-service
network can make such applications available to every employee
...
Voice and data communications place distinctly different
demands on the network
...
Data does not
...
So networks that transmit all three must be managed to meet the
disparate requirements of data and voice/video
...
These techniques include the
strategic placement of VLANs, and the use of Class of Service (CoS) packet marking and Quality of
Service (QoS) network mechanisms
...

Professional consulting services are available through the Avaya Communication Solutions and
Integration (CSI) group
...
This assessment helps to prepare a customer’s
network for IP telephony, and also gives critical network information to Avaya support groups that will
later assist with implementation and troubleshooting
...


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Avaya IP Telephony Implementation Guide
Table of Contents
1

Introduction to VoIP and Avaya Products
...
1
...
7
Servers
...
7
Stations
...
7
1
...
Avaya Server-Gateway and Trunk Architectures
...
8
IP-enabled DEFINITY System
...
10
S8500 Media Server
...
11
S8300/G700/G350/G250
...
12
IP-Connect with Remote G700/G350/G250 Gateways
...
14
1
...
VoIP Protocols and Ports
...
16
2
...
General Guidelines
...
16
Speed/Duplex
...
2
...
18
Calculation
...
19
WAN Overhead
...
19
L2 Fragmentation
...
3
...
20
General
...
20
802
...
21
Rules for 802
...
21
DSCP
...
24
QoS on a Router
...
25
Traffic Shaping on Frame Relay Links
...
27
Avaya IP Telephony Implementation Guide

4

3
...
S87xx/S8500 Servers
...
27
S87xx/S8500 802
...
28
3
...
S8300 Server
...
3
...
29
G700 P330/C360 L2 Switch
...
29
G700 802
...
30
G700 in Octaplane Stack vs
...
30
G350 Media Gateway
...
31
General Guidelines Related to Gateways
...
4
...
32
C-LAN Capacity and Recommendations
...
33
C-LAN and MedPro/MR320 Network Placement
...
33
C-LAN and MedPro/MR320 802
...
34
MR320 Capabilities and MR320 Bearer Duplication
...
35
IP Server Interface (IPSI) Board
...
5
...
36
ethernet-options
...
36
ip-interface
...
37
ip-codec-set
...
38
ip-network-map
...
41
trunk-group and signaling-group
...
43
system-parameters mg-recovery-rule
...
43
SAT Troubleshooting Commands
...
46
4
...
Basics
...
Current Models
...
47
DHCP Lease Duration
...
48
Boot-up Sequence
...
49
Keepalive Mechanisms
...
2
...
51
IP Phone and Attached PC on Same VLAN
...
52
4
...
Gatekeeper Lists and DHCP Option 176
...
54
Branch Site
...
55
Verifying the Gatekeeper Lists
...
57
Appendix B: Cisco Auto-Discovery
...
65
Appendix D: Access List Guidelines
...
69
Appendix F: Sample QoS Configurations
...
75
Appendix H: IPSI Signaling Bandwidth Requirements
...
80

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1

Introduction to VoIP and Avaya Products

This section provides a foundation to build upon for the rest of this document
...

1
...
From servicing a simple call to making
complex decisions associated with large contact centers, the call server is the primary component of an IP
telephony system
...

The following are some common terms for a call server
...

- Call Server – generic term
- Call Controller – generic term
- Gatekeeper – H
...
248 term
Gateways
A gateway terminates and converts various media types, such as analog, TDM, and IP
...

The following are some common terms for a gateway, and they are generally used throughout the
industry
...
323 term
- Media Gateway – H
...

Stations
There are several technical terms for what most would call a telephone, and some that are generally used
throughout the industry are listed below
...
323 general term that includes IP phones and other endpoints
- Terminal – H
...

Trunks
Trunks connect independent telephony systems together, such as PBX to PBX, or PBX to public switch,
or public switch to public switch
...
IP telephony introduces another trunk type –
the IP trunk
...
323
...


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Avaya IP Telephony Implementation Guide

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1
...
Also included in the diagrams are the protocols used to
communicate between the various devices
...
They house port boards, which, among other things, terminate stations
and trunks
...
)
The DEFINITY® call servers are the processor boards inserted into the processor port network
(PPN)
...

The port networks are connected together via a port network connectivity (PNC) solution, which can
be TDM-based (Center Stage PNC) or ATM-based (ATM PNC)
...

Control Channel Message Set (CCMS) is the Avaya proprietary protocol used by the DEFINITY
servers to control the port networks (cabinets and port boards)
...
225 - RAS &
Q
...
225

MedPro
RTP

C-LAN

EPN

to P

Enterprise
IP Network

TDM bus
SCC

CCMS and bearer
over TDM or ATM

MCC

Center Stage
or ATM PNC

MCC

DCP
STN

MCC
STN

DCP

RTP
audio

MedPro

TDM bus

Analog

EPN

to P

IP Net

PPN

Procr

C-LAN

CCMS from processor
to port boards across
backplane

Procr

EPN

Analog

Figure 2: IP-enabled DEFINITY System

-

KW

IP-enabled DEFINITY System is the same architecture as before, but with IP port boards added
...

H
...
323, is the protocol used for call signaling
...
225 itself has two
components: RAS for endpoint registration, and Q
...

The IP Media Processor (MedPro) board is the IP termination point for audio
...
0 there is a higher capacity version of the MedPro board called IP Media Resource 320
(MR320)
...
These boards perform the conversion between TDM and IP
...


Avaya IP Telephony Implementation Guide

9

Multi-Connect
Adjunct Location

Medium/Large Enterprise - Main Location

L2 switch
IP

IP

s8700

L2 switch

CCMS over
TCP/IP

H
...
225 - RAS &
Q
...

Port networks get IP Server Interface (IPSI) boards to communicate with the S87xx call servers
...

Not all port networks require IPSI boards, because Center Stage PNC and ATM PNC are still present
to connect the port networks
...
The S8500 gives the
same call processing capability without the redundancy and added reliability of duplicated servers
...


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Avaya IP Telephony Implementation Guide

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IP-Connect
-

Enterprise
IP Network
CCMS over
TCP/IP
H
...


-

G650/G600 media gateways
still use C-LAN and
MedPro/MR320 boards, as
well as the other traditional
port boards used in the MCC1
and SCC1
...


-

s8700

With IP-Connect the
traditional port networks –
MCC1 and SCC1 – are
replaced with new, 19-inch
rack-mountable Avaya G650
or G600 Media Gateways
...
225
IP

g700 with
s8300 ICC
VoIP mod

IP

H
...
248 protocol
...

(Refer to current product offerings for exact specifications
...
The G700
supports IP routing and IP WAN connectivity with an
expansion module, and the G350 and G250 support them
natively
...
The G350 and G250 are built on a
new IP platform, also with similar CLI
...

The VoIP media module serves the same function as the
MedPro board
...

The Avaya S8300 Media Server in internal call controller (ICC)
mode is the call server
...

The S8300 is not front-ended by C-LANs; it terminates the call
signaling natively
...
225

H
...
931 signaling
Control
IP Network

IP

IP

IP

backup
H
...
225
IP Net

PN

IPSI
IPSI

PN

IPSI

IPSI

CCMS
RTP
audio

PN

C-LAN

Center Stage
or ATM PNC

Enterprise
IP Network

MCC

IP

IP

DCP

ST
N

MCC

g700 with
VoIP mod
DCP mod
Analog mod
T1/E1 mod

RTP

Analog

to
P

CCMS and bearer
over TDM or ATM

g350 with
VoIP mod

TN

SCC

H

g700 with
VoIP mod

trol
N
con
WA
way
ate
ia g
med
248

...

The remote S8300 is in local survivable processor (LSP) mode to take over as call server if
connectivity to the S87xx servers is lost
...
225

IP

backup H
...
248

IP

WAN

IP
IP

STN

g
ia

y
wa
ate

co

ol
ntr

g350 with
VoIP mod

g700 with
VoIP mod
DCP mod
Analog mod
T1/E1 mod

RTP
IP

IP

G650
Analog

DCP

DCP

Analog

to
P

to P

d
me
48
2
H
...
225

IPSI
C-LAN
MedPro

g700 with
s8300 LSP
VoIP mod

IP

ST
N

G650

Remote Office

Figure 8: IP-Connect with remote G700/G350/G250s

-

KW

Remote gateways and stations are controlled by the S87xx servers via the C-LAN boards
...


Avaya IP Telephony Implementation Guide

13

Trunks
QSIG
H
...
931)
IP

DCP

Call
Manager

S8300 / G700

H
...
323 (Q
...
931
PRI
G650

H
...
931
T1 OR
PRI

Q
...
931
PRI

DEFINITY System

Figure 9: Trunks

This figure illustrates a broad picture to put trunks into context
...
This protocol is not relevant to
private, enterprise telephony systems
...
931 signaling
...
931 signaling
...

- QSIG is a standard, feature-rich signaling protocol for private systems, and it “rides on top of” Q
...
DCS is the Avaya proprietary
equivalent to QSIG, which also rides on top of Q
...

- Gatekeepers, such as the S8700, S8300 and S8500, and Cisco Call Manager in this illustration, can
connect to one another using IP trunks
...
323, but
Q
...
323 that does the call signaling
...
931
...

Generally speaking, traditional telephony systems support a broad range of QSIG features, while pure IP
telephony systems support a very limited range
...


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Avaya IP Telephony Implementation Guide

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1
...
The contents of the upper-layer
protocol messages are important to those who develop VoIP applications
...

Instead, they are concerned primarily with the transport mechanism – TCP and UDP ports – so that they
can verify and filter these message exchanges
...
323

Audio CODEC
G
...
729

H
...


H
...
225
RAS
Registration

Q
...
323 is the prevalent VoIP protocol suite
...

- H
...
931) component of H
...

- H
...

- H
...

- H
...
323
...

- G
...

- G
...

Encoded audio is encapsulated in RTP (real-time protocol), then UDP, then IP
...

- On Avaya solutions the UDP ports used to transport RTP streams are configured on the call
server
...

H
...
It is transported via TCP with port 2945 (1039 if
encrypted) on the media gateway controller side
...
It
is transported via TCP with port 5010 on the port network (IPSI board) side
...

2
...
Ethernet collisions – a major contributor to delay and jitter – are virtually eliminated
on switched networks
...
This
provides a cleaner design where VoIP hosts are not subjected to broadcasts from other hosts, and where
troubleshooting is simplified
...
When a PC is attached to an IP telephone, even if they are on separate VLANs, both PC
and phone traffic (including broadcasts) traverse the same uplink to the Ethernet switch
...

High broadcast levels are particularly disruptive to real-time applications like VoIP
...
Therefore, a
subnet/VLAN with only these Avaya hosts has a very low broadcast level
...
Case 1 is
one of the reasons for the recommendation to use separate voice subnets/VLANs
...
For this reason Avaya IP phones are designed to be very resilient
against broadcasts, with lab tests showing the phones operating satisfactorily even with 3,000 to 10,000
broadcasts per second, depending on the model
...
The recommended maximum
broadcast rate is 500 per second, and the absolute maximum is 1000 per second
...
Use 100M links, take
measures not to exceed the recommended maximum broadcast rate (500/s), and do not exceed the
absolute maximum broadcast rate (1000/s)
...

- Separate voice VLANs
...

- No more than ~250 hosts (/24 subnet mask) each on voice and data VLANs if IP phones have PCs
attached to them
...

- As low a broadcast rate as possible – 500/s recommended max; 1000/s absolute max
...
They are meant to provide the simplest
configuration by removing unnecessary features
...
When a device is first connected (or re-connected) to a port running STP, the
port takes approximately 50 seconds to cycle through the Listening, Learning, and Forwarding states
...
Enable a fast start feature on these ports to put them into the Forwarding state almost
immediately
...
If this feature is not
available, disabling STP on the port is an option that should be considered
...

Enable Rapid Spanning Tree and configure host ports as edge ports – As the name implies, Rapid
Spanning Tree Protocol (RSTP) is a faster and more advanced replacement for STP
...
Even if they don’t, there are
ways to combine RSTP and STP (depending on the network equipment), though certainly not as clean
as having RSTP throughout the L2 domain
...

Disable Cisco features – Cisco features that are not required by Avaya endpoints are auxiliaryvlan
(except for IP phones in a dual-VLAN setting as described in appendices A and B), channeling, cdp,
inlinepower, and any Cisco proprietary feature in general
...
The CatOS command set port host
automatically disables channeling and trunking, and enables portfast
...
For dual-VLAN IP telephone
implementations, see Appendices A and B for more information and updates regarding auxiliaryvlan
and trunking
...
1Q trunking on Cisco switches – If trunking is required on a Cisco CatOS
switch connected to an Avaya device, enable it for 802
...
This causes the port to become a plain 802
...
When trunking is not required, explicitly disable it, as the
default is to auto-negotiate trunking
...
There is a
significant amount of misunderstanding in the industry as a whole regarding the auto-negotiation
standard
...
This means that if a device with
fixed speed and duplex is connected to a device in auto-negotiation mode, the auto-negotiating device can
sense the other device’s speed and match it
...
Therefore, the auto-negotiating device always
goes to half duplex in this scenario
...
It is imperative that the speed and duplex
settings be configured properly
...
Suitable
for user PC connections, but not suitable for server connections or
uplinks between network devices
...
Not suitable for
multiple VoIP calls, such as through a MedPro/MR320 board
...
Device1 senses the speed and matches accordingly
...

10/half stable
...

Device1 senses no duplex negotiation, so it goes to half duplex
...
Device1 senses the speed and matches accordingly
...


Avaya IP Telephony Implementation Guide

17

100/full

100/full

10/half
100/half

100/full stable
...

Stable at respective speed and duplex
...


10/half
100/half

Table 1: Speed/duplex matrix

Layer 1 (L1) errors such as runts, CRC errors, FCS errors, and alignment errors often accompany a
duplex mismatch
...
The show port command on
Catalyst switches gives this information
...
The Avaya P330 switch
command is show rmon statistics
...
2

Bandwidth Considerations

Calculation
Many VoIP bandwidth calculation tools are available, as well as pre-calculated tables
...
The per-call rates for G
...
726 and G
...

- Voice payload and codec selection – The G
...
Since the audio is
encapsulated in 10-ms frames, and there are 100 of these frames in a second (100 * 10ms = 1s), each
frame contains 640 bits (64000 / 100) or 80 bytes of voice payload
...
726 codec payload rate is
32000bps and the G
...
This equates to 320 bits or 40 bytes and 80
bits or 10 bytes per 10-ms frame respectively
Voice Payload
G
...
726
G
...
number of frames

Packet size and packet rate – Because the voice payload rate must remain constant, the number of
voice frames per packet (packet size) determines the packet rate
...


-

Packet Rate
G
...
726
G
...
packet size

-

IP, UDP, RTP overhead – Each voice packet inherits a fixed overhead of 40 bytes
...
Add up the voice payload and overhead per packet, and multiply
by the number of packets per second
...
711 and a G
...
(Remember that there are 8 bits per byte
...
711: (160B voice payload + 40B overhead)/packet * 8b/B * 50 packets/s = 80kbps
G
...
729: (20B voice payload + 40B overhead)/packet * 8b/B * 50 packets/s = 24kbps
The calculations above do not include the L2 encapsulation overhead
...

L2
header

IP
20 B

UDP
8B

RTP
12 B

Voice Payload
Variable

L2
trailer

Figure 12: L2 overhead

Ethernet Overhead

G
...
4kbps
G
...
6kbps
Ethernet has a header of 14 bytes and a trailer of 4
G
...
4kbps
bytes
...
726 30-ms call over Ethernet = 49
...
729 20-ms call over Ethernet = 34
...
729 30-ms call over Ethernet = 25
...
Nevertheless, the preamble and SFD
consume bandwidth on the LAN, so the total
Ethernet overhead is 26 bytes
...
4kbps (26 * 8 * 50), which must be added to the 80kbps for G
...
726, and 24kbps for
G
...
For full-duplex operation the totals are 90
...
711, 50
...
726, and 34
...
729
...

WAN Overhead
The WAN overhead is calculated just like the Ethernet overhead, by determining the size of the L2
encapsulation and figuring it into the calculation
...
For example, the
PPP overhead is only 7 bytes
...
6kbps (14 * 8 * 50), assuming 20-ms voice
packets
...
729 20-ms call over PPP = 26
...
726 and 30kbps for G
...

G
...
8kbps
Significant bandwidth savings are realized by using a
compressed codec (G
...
729 20-ms call over 14-B L2 = 29
...
Note that in today’s data networks most, if
G
...
6kbps
not all, WAN links are full duplex
...
The first factor, maximum transmission unit (MTU), involves
fragmenting the layer 3 (L3) payload
...
711 and 60 bytes for G
...
If the
MTU on an interface is set below these values the IP payload (UDP + RTP + voice payload) must be
fragmented into multiple IP packets, each packet incurring the 20-byte IP overhead
...
The 20-ms G
...
The 180-byte
payload must be divided into three fragments of 80 bytes, 80 bytes, and 20 bytes
...
A single 200-byte IP packet
must be fragmented into three separate IP packets to meet the 100-byte MTU
...

MTU should not be an issue for VoIP because most interfaces have a default MTU of 1500 bytes
...
Even if the MTU is not set to a level
that would fragment VoIP packets, the principle of fragmenting the L3 payload and incurring additional
L3 and L2 overhead applies universally
...
A low MTU value, relative to any given IP packet size, always increases
L3 and L2 overhead as illustrated with the VoIP example
...

L2 Fragmentation
The second factor involves fragmenting the L2 payload, or the entire IP packet
...
For example, the fixed cell size for ATM is 53 octets (bytes), with 5 octets for ATM
overhead and 48 octets for payload
...
Therefore, the L3 packet (not just the IP payload, but the entire IP packet) is
fragmented and carried inside multiple ATM cells
...
711 IP packet would require five ATM
cells (25 octets of ATM overhead), whereas a 60-byte G
...
Refer to Appendix C for information regarding RTP header compression
...
The router could pay a significant processor penalty if the compression is
done in software
...
This is because
most carriers (ATT, Verizon, Sprint) convert Frame Relay to ATM for the long haul, between the local
central offices
...
5 and FRF
...
In this process the Frame Relay header is translated to an ATM
header, and the Frame Relay payload is transferred to an ATM cell
...
Therefore, it is beneficial to limit the size of the voice packet even
when the WAN link is Frame Relay
...
3

CoS and QoS

General
The term “Class of Service” refers to mechanisms that mark traffic in such a way that the traffic can be
differentiated and segregated into various classes
...
If an endpoint marks its
traffic with L2 802
...
The fact that
certain traffic is marked with the intent to give it higher priority does not necessarily mean it receives
higher priority
...

CoS
802
...
These mechanisms are supported by the IP telephones and most IP port boards
...
Although TCP/UDP source and destination ports are not
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Avaya IP Telephony Implementation Guide

20

CoS mechanisms, they are inherently used to identify specific traffic and can be used much like CoS
markings
...

802
...
1Q tag and its insertion point in the Ethernet and 802
...

Note that in both cases the 802
...
3 frames
...
1Q tagged frames
...
The Tag Protocol Identifier (TPID) field has hex
value x8100 (802
...
This value alerts the switch or host that this is a tagged frame
...
1Q tagging, the TPID field is mistaken for the Type or Length
field, which results in an erroneous condition
...
3 "Ethernet" frame
Dest Addr
(6 octets)

Len
(2)

802
...
2
LLC

Src Addr
(6 octets)

802
...
1p/Q
Priority
(3 bits)

VLAN ID
(12 bits)

CFI

TPID
(2 octets)

TCI
(2 octets)

Figure 13: 802
...
The Priority field is the “p”
in 802
...
(“802
...
1Q tag has significance
...
1Q was used
primarily for VLAN trunking, and the Priority field was not important
...

Rules for 802
...
Is tagging required to place the frame on a specific VLAN (VLAN tagging)?
2
...

1
...

- On a single-VLAN port there is no need to tag to specify a VLAN, because there is only one
VLAN
...
1Q standard specifies the use of VID 0
...
” Some Ethernet switches do not properly interpret VID 0, in
which case the port/native VID may need to be used, but this is not the standard method
...
This treats all incoming traffic
on that port as high-priority traffic, based on the configured level
...
The low-priority
device (PC) would not tag at all
...

2
...

- A multi-VLAN port has a single port/native VLAN and one or more additional tagged VLANs,
with each VLAN pertaining to a different IP subnet
...

- For the attached device that belongs on the port/native VLAN, follow the points given for rule 1
above
...

- An attached device that belongs on any of the tagged VLANs must tag with that VID and the
desired priority
...
In this case the PC would send clear frames,
and the IP telephone should tag with the designated VID and desired priority
...
1Q tag, and many must
be explicitly configured to receive it
...
Avaya switches accept VID 0 without any special configuration, but some
Ethernet switches do not properly interpret VID 0
...
The following table shows the results of some testing performed by
Avaya Labs on Cisco switches
...
1(2)
Catalyst 4000 w/
CatOS 6
...
0(5)WC2
Conclusion

Accepted VID 0 for the native VLAN when 802
...

Would not accept VID 0 for the native VLAN
...
Bug
ID is CSCdr06231
...
1Q trunking and tag with
native VID instead of 0
...
1Q trunking was disabled
on the port
...

Table 4: Sample VID 0 behaviors for Cisco switches

See Appendix A for more information on VLANs and tagging
...
The ToS field contains
three IP Precedence bits and four Type of Service bits as follows
...
DSCP utilizes the first six bits of
the ToS field and ranges in value from 0 to 63
...

8-bit Type of Service field
IP Precedence bits
Type of Service bits
0
1
2
3
4
5
DSCP bits
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Avaya IP Telephony Implementation Guide

6
0

0
7
0
23

Figure 16: Compare DSCP w/ original ToS

Ideally any DSCP value would map directly to a Precedence/ToS combination of the original scheme
...

On any device, new or old, having a non-zero value in the ToS field cannot hurt if the device is not
configured to examine the ToS field
...
These legacy devices (network and endpoint) may
contain code that only implemented the IP Precedence portion of the original ToS scheme, with the
remaining bits defaulted to zeros
...
For example, if an endpoint is marking with DSCP 40, a legacy network
device can be configured to look for IP Precedence 5, because both values show up as 10100000 in the
ToS field
...

Another hurdle is if the legacy code implemented IP Precedence with only one ToS bit permitted to be set
high
...
When these mismatches occur, the legacy device may reject the DSCP-marked IP packet or exhibit
some other abnormal behavior
...

QoS on an Ethernet Switch
On Avaya and Cisco Catalyst switches, VoIP traffic can be assigned to higher priority queues
...
The number of
queues and the technical sophistication of the queuing vary among switches, but in general the more
advanced the switch, the more granular the queuing to service the eight L2 priority levels
...
1p/Q priority tag and assign each class of traffic to a specific
queue, but only if this is a default feature or it is explicitly configured
...
This frees the endpoint from having to tag its traffic with L2 priority
...
Unlike
Ethernet switches, routers typically do not have a fixed number of queues
...
For example, Cisco routers have standard first-in first-out queuing (FIFO),
weighted fair queuing (WFQ), custom queuing (CQ), priority queuing (PQ), and low latency queuing
(LLQ)
...
Each queuing
mechanism behaves differently and is configured differently, but following a common sequence
...
Then the traffic
must be assigned to a queue in one of the queuing mechanisms
...
[2 p
...
For example, Cisco requires traffic shaping on Frame Relay and ATM links to
help ensure that voice traffic is allotted the committed or guaranteed bandwidth (see “Traffic Shaping on
Frame Relay Links” below in this section)
...
Serialization delay is the delay
incurred in encapsulating a L3 packet in a L2 frame and transmitting it out the serial interface
...
The concern is that large, low priority
packets induce additional delay and jitter, even with QoS enabled
...
The following matrix is taken directly from the “Cisco IP
Telephony QoS Design Guide” [2 p
...


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WAN
Link Speed
56 kbps
64 kbps
128 kbps
256 kbps
512 kbps
768 kbps

64 bytes
9 ms
8 ms
4 ms
2 ms
1 ms
640 us

128 bytes
18 ms
16 ms
8 ms
4 ms
2 ms
1
...
56 ms
5
...
24 ms

1500 bytes
214 ms
187 ms
93 ms
46 ms
23 ms
15 ms

Table 5: Cisco seralization delay matrix

Consult Cisco’s documentation for detailed information regarding traffic shaping and LFI, and be
especially careful with LFI
...
This is because a single L3 packet that was once transported in a single L2
frame, is now fragmented and transported in multiple L2 frames
...

Instead of implementing LFI, some choose to simply lower the MTU size to reduce serialization delay
...
As explained in section 2
...
Lowering the MTU is generally not
advisable and may not provide any added value, because it adds more traffic to the WAN link than LFI
...
One should have a thorough understanding of the traffic traversing the
WAN link before changing the MTU
...

However, it is on the router where QoS is needed most, because most WAN circuits terminate on routers
...
This appendix does not contain
configurations for all the issues discussed in this document, but it gives the reader a place to start
...
It is good practice to baseline the VoIP response (ie, voice quality) on a
network without QoS, and then apply QoS as necessary
...
If
voice quality is acceptable without QoS, then the simplest design may be a wise choice
...
Then QoS can be implemented on the LAN segments
as necessary
...
Simple
routing and switching technologies have been around for many years and have advanced significantly
...
518], “switching” being used as a generic term here), without heavy processor intervention
...
Many of the newer devices can
handle this additional processing in hardware, resulting in maintained speed without a significant
processor burden
...
5-18])
...
This can result in an
overall performance degradation from the network device, and even device failure
...
Since most QoS
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policies are implemented on WAN links, the following very general points for Cisco routers are offered to
increase the level of confidence that QoS remains in hardware
...

- Newer hardware platforms are required: 2600, 3600, 7200, and 7500
...
) are required: Consult Cisco to determine which hardware
revision is required for any given module
...

- Newer IOS is required: 12
...

Several things should be examined whenever QoS is enabled on a network device
...
It is likely that
the level will have gone up, but the increase should not be significant
...
The processor load must remain at a manageable level
(max 50% average, 80% peak)
...
There is no added value in leaving a particular QoS
mechanism enabled if VoIP response has not improved under stressed conditions
...

Traffic Shaping on Frame Relay Links
Experience to date supports Cisco’s requirement to use traffic shaping on frame relay links [2 p
...

Simply stated, VoIP traffic must be sent within the committed information rate (CIR) and not in the burst
range
...
Under this constraint one solution for maximizing bandwidth is to make the
CIR as large as possible, and this is dictated by the end of the PVC that has the smaller access circuit
...


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3

Guidelines for Avaya Servers and Gateways

This section gives guidelines for Avaya servers and gateways, and covers most of the IP-telephonyrelated configurations
...

Avaya Communication Manager is the call processing software that runs on Avaya servers, and it is
configured via the Switch Administration Terminal (SAT) interface
...

The Avaya Site Administrator (SA) is a client software application used to access the SAT interface on all
Avaya servers
...

3
...
S87xx servers
operate in a redundant pair, whereas the S8500 is a simplex server
...
The web interfaces are designed to facilitate all anticipated configuration and
management requirements, and there is little or no need for a customer to access the Linux shell
...
SAT administration is performed on
the active server, and it is automatically carried over to the standby server
...
If the two
servers are on the same subnet there is a virtual IP address, which is labeled the active server address in
the Configure Server – Configure Interfaces screen of the Maintenance Web Interface
...
If the two S8700 servers are not on the same subnet (server
separation), there is no virtual active server address
...

The S8700 SAT interface may be accessed using Avaya Site Administration (ASA) or by telnet-ing to
port 5023: telnet 5023
...
The standby server does not permit access to SAT
...
ASA
supports Secure Shell (SSH) access for system administration
...

SAT access to the S8500 is similar to that of the S87xx server pair, except that there is only one server
...
1 a S8500 main server or S8500 LSP supports Processor Ethernet (PE)
...
323 IP endpoints, H
...
An S8500 PE interface uses one of the native NICs on the server and allows for
direct connections to H
...
There
are, however, configuration limitations, which are defined in the Overview for AVAYA Communication
Manager, Document ID 03-300468, available on the support
...
com website
...
It is critical to configure the speed and duplex
correctly on the server interfaces used to communicate with the IPSI boards
...

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The web admin screen has a pull-down menu for the various speed/duplex settings
...
A “current speed” description
next to this pull-down menu indicates the current speed and duplex, but it does not indicate whether these
settings were manually configured or reached via auto-negotiation
...

- Start with the server and the Ethernet switch port set to auto-negotiate (default)
...
When in doubt always return to this
base state
...
With the Ethernet switch port still at autonegotiate, it should now show that the negotiated speed/duplex is 100/half
...

- Manually configure the Ethernet switch port to 100/full
...
Both sides are now optimally configured for 100/full
operation
...

- Following the instructions in section 2
...

S87xx/S8500 802
...
On this network there is no need to configure QoS, because
all traffic is port network control traffic and has equal priority
...
This does not apply on the control IP network
...
If QoS is desired and properly configured on this network, it may be
necessary to have the S87xx/S8500 server(s) tag/mark the port network control traffic
...
Traffic is tagged/marked from these interfaces on a per destination basis for each
IPSI board, as administered on the SAT ipserver-interface form (see section 3
...
For the 802
...
The interfaces that communicate with IPSI boards have this option, and
the others do not
...
3, heading “Rules for 802
...

3
...
The S8300 is similar
to the S87xx/S8500 in many ways
...

In a G700 the S8300 must have an IP address on the same IP subnet as the MGP, with the same mask and
default gateway (see G700 section below)
...
In a G350/G250 a VLAN must be designated as the ICC VLAN, and the S8300 must have
an IP address on the IP subnet pertaining to that VLAN (see G350 section below)
...
In ICC mode the S8300 is a standalone call server
...
An LSP does not accept station registrations or assume call processing
responsibilities until it becomes active, which occurs when a gateway registers to it
...
This could also be done by telnet-ing to the S8300 and
typing sat from the Linux shell
...

The S8300 connects to the G700/G350/G250 via a backplane 100M Ethernet interface, which is not
configurable
...
3

G700/G350/G250 Media Gateways

G700 P330/C360 L2 Switch
The P330 L2 switch is the base platform for the G700
...
The asynchronous port (9600/8/N/1)
marked CONSOLE on the face of the G700 connects the user to the P330 CLI
...

The most common ones are probably the WAN router module and the 16-port Ethernet module
...
There is also an Octaplane® slot on the
back of the chassis, just like the P330
...

Three components of the P330 should be configured: the inband management interface, the default
route, and the switch itself
...
The inband interface requires
a VLAN, an IP address, and a mask
...

Once configured, the inband interface should be thought of as a host attached to the P330
...

However, like most L2 switch management interfaces, the inband interface is associated with a specific
VLAN
...
Many mistakenly think that any host attached to the P330 should be able to access the
inband interface directly, and this is not necessarily true
...

Like any other IP host, the inband interface needs a default route if it is to route off of its VLAN/subnet
...
If there is more than one router on the inband VLAN/subnet, the inband
interface may have additional routes based on destination subnets or hosts
...

Finally, the P330 L2 switch itself has various configuration parameters, such as Spanning Tree, VLANs,
trunking, and speed/duplex
...

G700 Media Gateway Processor (MGP)
The media gateway processor (MGP) is the media gateway portion of the G700
...
These media modules include analog port modules for
analog phones, DCP port modules for DCP phones, DS1 modules for TDM trunks, and others
...
Each VoIP module is
equivalent to a MedPro board and has 64 audio resources
...

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Like the P330 inband management interface, the MGP should be thought of as a host on the P330 L2
switch
...
The MGP
requires a VLAN, IP address, and mask
...
The MGP may be on the same VLAN as the inband interface, or on a different
VLAN
...
Like the
inband interface, the MGP also needs at least a default route to route off of its VLAN/subnet
...

Each VoIP media module also requires an IP address using the set interface voip v# command
...
The internal VoIP module is voip v0
...
show mm
shows all the media modules and their slot numbers
...
1p/Q and DSCP
The G700 can receive its audio and call signaling priority values from the call server’s ip-networkregion form or from local configuration
...
The
command set qos control determines which set of values is used
...
If configured locally, the set qos commands are used to administer the
settings
...

G700 in Octaplane Stack vs
...
There are pros and cons to this
...
The cons are that the initial configuration can be a little more
complex, and a problem associated with the stack can adversely affect the G700
...

Device and uplink management are key factors
...
This allows the P330 components of
all the G700s to be managed via a single inband interface
...

When determining whether or not a single G700 should be added to an existing Octaplane stack of P330
switches, the relative importance of the G700 to the other devices is another factor
...
A P330 switch’s primary role is that of L2
switching – processing and forwarding Ethernet frames, managing broadcast domains (VLANs),
participating in Spanning Tree, etc
...
These points
are mentioned to provoke thought in design and implementation
...

Bandwidth is another key factor for using or not using the Octaplane stack
...
Each VoIP module consumes a maximum of approximately 6Mbps to service 64 G
...
With up to five VoIP modules on a single G700, the maximum bandwidth
consumption is approximately 30Mbps
...
Therefore, a single 100M uplink from EXT1 or EXT2
to another Ethernet switch is sufficient for the G700 itself
...
If the hosts on the expansion module are IP telephones, a 100M uplink is
sufficient
...

G350 Media Gateway
The G350 is similar to the G700 in many ways
...

Two significant differences between the G350 and G700 are capacity and architecture
...
As such, the G350’s internal VoIP
module has only 32 audio resources, as opposed to 64 in the G700’s internal VoIP module, and in the
external VoIP media module and MedPro board
...

The primary architectural difference between the G350 and G700 is that the G350 is an integrated
platform
...
In addition, a L3 router is integrated into the G350, whereas the G700 can accept a L3 router
as an expansion module
...
The L2 switch and MGP
commands are practically the same as on the G700, using set commands similar to the P330 switch and
Cisco’s CatOS
...
The G350 utilizes both command sets in a single CLI
...
One of the G350’s IP interfaces must be designated the primary
management interface (PMI)
...
The PMI, among other things, is the interface used by the
MGP and internal VoIP module
...
248 media gateway signaling and RTP audio are
sourced by and terminated on the PMI
...
This is done by inserting the command iccvlan under the desired VLAN interface
...
Refer to the G700 sections above
for more details on each subject
...

But there is a single default route for the whole unit, as opposed to a route for the P330 component
and a route for the MGP component
...

- The G350 and G700 L2 switch configurations (Spanning Tree, VLANs, trunking, speed/duplex) are
very similar
...
1p/Q and DSCP configurations are very similar
...

G250 Media Gateway
The G250 is very similar to the G350, but with less capacity – ~10 users max
...

Like the G700 and G350, the G250 is capable of housing an S8300 server in ICC or LSP mode
...
323 gatekeeper
...
For simplicity, SLS can be thought of as
an integrated LSP with very limited features
...

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General Guidelines Related to Gateways
The MGC List, Transition Point, and Primary Search Time must be configured properly on all gateways
...
There may be undesirable behaviors if these parameters are not configured
properly, such as a gateway registering with an LSP when a primary controller is available; a gateway
registering with an LSP too soon after an outage; and different gateways at the same location registering
with the LSP at different times
...
avaya
...

When connecting a gateway to another Ethernet switch, the uplink between the two switches should be
fixed at 100M/full-duplex (see section 2
...
Furthermore, if 802
...
1Q trunking enabled with matching VLANs on the connected ports
...
If the LSP, media gateway, and IP phones are on different subnets, they
depend on a router or routing process to function
...
The IP telephony system should be able to function even if that
router or routing process fails
...

For remote offices where the WAN link terminates on the gateway, whether on the X330WAN router or
natively on the G350/G250 itself, the DSCP values for audio and signaling must be 46 and 34
respectively
...
These values can be configured locally via the set qos
bearer/signal commands, or they can be downloaded from Communication Manager
...
A gateway is configured to use the Communication Manager values by
executing set qos control remote on the MGP/gateway CLI
...

The G700, G350, and G250 gateways are all administered in Communication Manager via the SAT
media-gateway form, which is covered in section 3
...

3
...
248 media gateways
...
248 protocol
...
The most
significant boards related to IP telephony are the C-LAN (TN799DP), MedPro (TN2302AP), and MR320
(TN2602AP) boards
...

C-LAN Capacity and Recommendations
The Control LAN (C-LAN) board is the IP interface for many functions, including H
...
248 media gateway control signaling, connectivity to various adjuncts,
and SAT administrative access via TCP/IP
...
The C-LAN board can support over 400 sockets under light usage
conditions
...

Furthermore, regardless of usage it is highly discouraged to operate the C-LAN near maximum capacity
in a production environment
...

- Adjuncts such as CMS, CDR, AUDIX Messaging System, and others should be placed on separate
C-LAN boards that are not used for call signaling or media gateway control signaling
...

In a typical call center environment, design for a normal operating load of 200-250 IP stations plus 6
media gateways per C-LAN
...

The number of signaling groups (IP trunks) per CLAN depends greatly on the configuration and
usage of each signaling group
...
The greater the usage of the signaling group (frequency of
calls, features utilized during calls, number of simultaneous calls, etc
...
As a very general rule based on anecdotal evidence of typical IP trunk usage,
and assuming calls share IP signaling connection, substitute one signaling group for ten IP stations in
the two preceding bullet items
...


C-LAN and MedPro/MR320 Protocols and Ports
Call signaling and media conversion between analog, TDM, and IP are key IP telephony functions
...
The following table lists the
protocols and ports used by both boards
...
5, heading “ip-network-region” gives instructions on
how to configure the MedPro/MR320 UDP port range
...

UDP 1719
TCP 1720
C-LAN

MedPro/
MR320

TCP 2945
TCP 1039
UDP 2048 – 65535 (configurable)

H
...
225 Q
...
248 media gateway control signaling
H
...
Keep in mind that both call signaling and audio
from all IP endpoints require these boards
...
On the other hand, a server farm is typically where
the most reliable and redundant network resources are deployed
...

C-LAN and MedPro/MR320 Speed/Duplex
Use the SAT ip-interface form to configure the speed and duplex for the C-LAN and MedPro/MR320
boards
...
This results in
much better system stability and audio quality than if the boards and Ethernet switch ports are left to autonegotiate
...
1 under the “Speed/Duplex” heading
...
The default
speed/duplex setting on the TN799DP C-LAN board is 10/half, to make it backwards compatible with the
previous TN799C board, which could only do 10/half
...
If for any reason a board loses this programming, it reverts back to
the board’s default
...
8Mbps, which is what is required for 64 G
...
8Mbps for 64 G
...

ms calls over Ethernet
...
Therefore, the
The maximum MR320 throughput is
minimum speed/duplex requirements are 100/half for
29Mbps for 320 G
...

the MedPro and 10/half for the C-LAN
...

If there is poor audio quality on calls going through a particular MedPro/MR320 board, follow these steps
to determine if a speed/duplex mismatch between the MedPro/MR320 and the Ethernet switch is the
cause
...

- Check for L1 errors as instructed in section 2
...

- Send a continuous ping (ping -t) to the MedPro/MR320 from a Windows machine
...

C-LAN and MedPro/MR320 802
...
5, headings “ip-interface” and “ip-network-region
...

MR320 Capabilities and MR320 Bearer Duplication
The TN2602AP IP Media Resource 320 provides either 80 or 320 encrypted or unencrypted channels of
2-way audio RTP streams or conversations
...
The
MR320 supports G
...
729A/B and G
...
See Table 6 for a comparison of Medpro and
MR320 capabilities
...
As result,
the port network can have either two duplicated TN2602AP circuit packs or two load balancing
TN2602AP circuit packs, but not both a duplicated pair and a load-balancing pair
...

The TN2602AP IPMedia Resource 320 can provide duplicated bearer for IP Connected Port Networks
...
A port network supports a
maximum of two TN2602AP circuit packs and they can be administered for duplication
...
State of health parameters exist
between the two boards to determine when it is appropriate to interchange duplicated TN2602AP circuit
packs
...
Duplicated TN2602AP circuit packs in a PN share a virtual IP and virtual
MAC address
...
In addition to the
virtual IP address, each TN2602 has a "real" IP address
...

Whichever TN2602AP circuit pack is active is the recipient of those packets
...
State-of-health, call state, and encryption information is shared between TN2602s
during this negotiation
...
It is also possible to invoke an interchange manually via a software command
...
In addition, the Ethernet switch or
switches that the circuit packs connect to must also be in the same subnet
...
This identification process provides a consistent virtual interface for calls
...
In addition, both circuit packs must have the latest firmware that
supports bearer duplication
...
711 - 80 or 320 channels by license,
channels, 48 maximum encrypted
unencrypted or encrypted
channels
...
729A, G
...
729B and G
...
1 32 maximum
channels by license, unencrypted or
unencrypted, 24 maximum encrypted
encrypted
channels
...
726A - 80 or 320 channels by
license, unencrypted or encrypted
...
38/Modem relay
Fax/Modem Pass-thru
TTY Relay
TTY Pass-thru
Echo Tail
SSH/SCP Support
Active-Standby Failover

16 unencrpyted, 12 encrypted
64 G
...
729
32 G
...
729 unencrypted, 24 encrypted
32 ms tail
No
No

● G
...
1, because it is a last-resort measure
...
This is
manifested by bridge join/leave messages for CatOS-based switches, and interface up/down messages for
IOS-based switches
...
Sometimes this
problem is a compatibility issue between the MedPro and the Cisco switch
...
1, headings “Ethernet Switches” and “Speed/Duplex” have been followed, if the Cisco switch
port continues to flap up and down, consider the options described in the next paragraph
...
6]” describes the flapping problem mentioned above and offers a suggestion to
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adjust the jitter tolerance (not related to audio jitter) on Cisco switches
...
This command increases the jitter
tolerance to 3
...
4-nsec default
...
This adjusts the carrier transition delay to 4 seconds
...

IP Server Interface (IPSI) Board
The IP Server Interface (IPSI) board is installed in a G650/G600, MCC1, or SCC1 port network, and it is
the port network’s interface to communicate with the call server(s)
...

If IP Control is ‘y’ the board is acting as an IPSI; otherwise (‘n’) it is acting as a tone clock
...
Ignore
Connectivity in Server Arbitration has to do with whether or not connectivity to this IPSI is factored into
the decision to interchange S87xx servers
...
The intent would
be to avoid server interchanges caused by frequent and inconsistent loss of communication to this IPSI
...
Host is the board’s static IP address if configured manually, or the hostname
if the address was obtained via DHCP
...
Socket Encryption, if the parameter is
present, allows the control link between the IPSI and call server to be encrypted
...
1p and DiffServ parameters contain the values to be applied to the call server when
communicating with this IPSI board (values are not applied to the IPSI board itself)
...
From the IPSI board type ipsilogin at the [IPSI]: prompt, and enter the login name and password
to access the [IPADMIN]: prompt
...
The commands
to display and configure the L2 and L3 priority values are show qos, set vlan tag, set vlan priority, and
set diffserv
...
3,
particularly the heading “Rules for 802
...

3
...
This section covers
the forms used to configure general IP telephony
...
Some also have a list option to view, for
example, a broad list of stations without seeing in detail how each station is configured
...
0 each IP board’s speed and duplex settings are configured using
the ip-interface form
...
With
each new system or IP board installation, one standard procedure should be to apply matching
speed/duplex settings to each IP board and its corresponding Ethernet switch port
...
This form is used to define arbitrary names and associate an IP address
with each name
...
168
...
10, and the name “medpro_80” could be defined to describe a MedPro
board on the 80 subnet with address 192
...
80
...

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ip-interface
Options are change, display, and list
...
The first step
is to associate a board Type and Slot # to a previously defined Node Name, and to give the board a Subnet
Mask and default Gateway and assign it to a Network Region
...
This assigns the IP address
192
...
80
...
Then the board can be given the mask 255
...
255
...
168
...
254
...

802
...
A number (including 0) in the
VLAN field indicates the VID, and it means that tagging is enabled on the board with that VID
...
The letter ‘n’ in this column means that tagging is disabled on the board, and a blank means that
tagging is not supported on the board
...
3 under the heading “Rules for 802
...
”
The speed and duplex settings for an IP board are configured on this form under the Ethernet Options
heading
...
While this parameter is configurable, it is restricted by licensing
...

The 2602AP MR320 board can be administered for Critical Reliable Bearer
...
The duplicated MR320s also share a virtual
MAC address that is automatically assigned by one of four virtual MAC tables
...

The C-LAN board parameter Number of CLAN Sockets Before Warning
...
4, heading “C-LAN Capacity and Recommendations
...
Although the recommended number of sockets on a C-LAN may be less than 400, it is advisable
in many cases to wait until 400 (default value) to trigger an alarm
...
The default
value should only be changed by AVAYA Services
...
323 Endpoints and the Allow H
...
The Gatekeeper Priority parameter is used for Alternate Gatekeeper lists and is available when
H
...
The lower the number the greater the priority
...
This form is used to assign an extension (required for call
processing) to a C-LAN board, and to specify other parameters
...
The Type is Ethernet
...
The Link number can be any available number
from the output of the display communication-interface links command
...

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ip-codec-set
Options are change, display, and list
...
Up to 7 codec sets may be defined with 5 codecs, listed in order of preference,
in each set
...
711 (uncompressed) and G
...
No silence suppression and 20-ms voice packets are also recommended
...
726A codec type but it is only available on
the MR320 (TN2602)
...
726A
...
The G
...
Rather than
not transmitting during silence, this codec transmits silence in a condensed format that requires less
bandwidth
...
729B is still
noticeably inferior to G
...

Larger packet size = less bandwidth
Note about voice packet size: Audio is encoded in
Smaller packet size = more bandwidth
increments called frames, with the typical frame size
being 10ms
...
Experience has shown that a 20-ms packet
Smaller packet size → high loss, low jitter
is a good compromise between audio quality and
network
bandwidth consumption
...
Going beyond 20ms reduces the number of packets put onto the
network, but there is greater potential for poor audio quality when there is high packet loss
...
Smaller packets work better in high loss, low
jitter networks
...

The Media Encryption portion of this form is an ordered list of preferred media encryption options
...
This list may
contain one or more items
...

For information on the remaining FAX, Modem, TDD/TTY, and Clear-channel parameters, see the
product documentation “Administration for Network Connectivity for Avaya Communication Manager”
(555-233-504), chapter 3, heading “Administering FAX, modem, TTY, and H
...
” See also the document “Avaya FoIP, MoIP, & TTYoIP” at www
...
com
...
This form is used to define the characteristics of an Avaya
Communication Manager network region
...
avaya
...

The Location parameter is used to assign IP stations in this network region to a specific geographic
location identifier
...

The Name is an arbitrary string to describe the network region
...

The UDP Port Min/Max is the range used for RTP audio by the MedPro and MR320 boards and VoIP
media modules in this network region
...

- 2048 is the beginning of the range by default, but this can be changed to a higher starting point
...
A starting port of 50000 is
outside the range of any reserved ports
...
711 codec) or 32 compressed audio streams
(G
...
The MR320 supports up to 320 audio streams, depending on licensing
and configuration
...

Therefore, to support X audio streams the UDP port range must contain 2X consecutive ports,
beginning with an even port and ending with an odd port
...
Duplicated Media Resource 320
(MR320) boards need 320x4 UDP ports or 1280 ports
...


The DiffServ (DSCP) and 802
...
The L2 values
are only applied to boards that have L2 tagging enabled via the ip-interface form
...

Ideally two different sets of L2/L3 values should be specified for signaling and audio
...
Appendix F gives examples of how the L3 values are used in conjunction with QoS on routers
...

Direct IP-IP Audio (shuffling) and IP Audio Hairpinning within a network region and across different
network regions are enabled and disabled on this form
...
If a feature that requires the media gateway, such as conferencing, is activated during the
call, the endpoints shuffle back to the MedPro/MR320 board or VoIP module
...

Hairpinning permits calls between IP endpoints to speak through the MedPro/MR320 board or VoIP
module, but without any transcoding
...
None of the Avaya IP telephones have this limitation
...
Also, for direct
IP-IP audio to function across different network regions, an inter-region codec set must be specified and
the regions must be connected via the inter-region connectivity matrix beginning on page 3 of this form
...
Since Avaya Communication
Manager 1
...
NAT has
been a hurdle for VoIP due to the fact that the address in the IP header is translated, but embedded IP
addresses in the H
...
This hurdle has been overcome to some extent with
the “NAT shuffling” feature in Communication Manager, without the need for H
...
See “NAT Tutorial and Avaya Communication Manager 1
...
avaya
...

Note: In addition to the ip-network-region form, shuffling and hairpinning must be enabled on two other
forms: the system-parameters features form, page 16; and the station form, page 2, for each station
...
Enabling this
feature causes the audio endpoints in this region to send periodic RTCP reports to VMM
...
The default server parameters
are configured on the system-parameters ip-options form
...

The RSVP feature requires careful integration with the IP network and must not be enabled without the
supporting IP network configurations
...
A better call admission control (CAC) mechanism is native to
Communication Manager as of 2
...
avaya
...

The H
...
See
the “Avaya Communication Manager Network Region Configuration Guide” at www
...
com
...
0
...
avaya
...
Related to IGAR is a new parameter on the cabinet form to
assign the cabinet to a network region
...
0, applies primarily to IGAR
...

ip-network-map
Options are change and display
...
If a station’s IP address does not fall into any of the
ranges configured on this form, the station is assigned to the same network region as the gatekeeper it
registers with
...
To understand how these methods of network region assignment
affect the station, see the “Avaya Communication Manager Network Region Configuration Guide” at
www
...
com
...
This field should only be used if DHCP option
176 is not available
...
The resulting functionality would be as follows
...

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IP phone registers with Communication Manager from data VLAN
...

Communication Manager directs phone to that voice VLAN
...

IP phone registers with Communication Manager from voice VLAN
...

Communication Manager directs phone to that voice VLAN, but phone is already on it
...
Using the recommended DHCP option 176
method, the phone applies the L2 priority values received from DHCP
...

station
Options are add, change, display, and list
...
To specify an IP station
the Type must be an IP model
...
This is changed to S##### – an automatically assigned internal port number – when the phone
registers with the call server
...
This field applies to non-IP stations as well, as an IP softphone can
take over an analog or DCP extension and emulate that set type
...
Direct IP-IP Audio and IP Audio Hairpinning for the individual
station is configured on page 2 of this form
...
These forms are used to define trunks, including H
...
This document is concerned only with the IP-specific configuration parameters
...
Once the members are
used for active calls the call server automatically changes the port designations to T#####, which are
internal port numbers
...

The signaling-group parameters are as follows
...
323
...

- Trunk Group for Channel Selection: Specify the trunk group configured as described above
...
323 signaling link, as defined in the local call server’s node-names ip and ip-interface forms
...
This is the default TCP port used by the gatekeeper for H
...

- Far-end Node Name: The node name of the far-end gatekeeper terminating the H
...

- Far-end Listen Port: 1720 by default if far-end gatekeeper is an Avaya server or Cisco Call Manager
...

- Far-end Network Region: The numeric identifier of the locally defined network region with which the
far-end gatekeeper is associated
...

- RRQ Required: ‘y’ if the signaling group is for a G150, R300, or MultiVOIP gateway
...


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-

-

-

-

Media Encryption: New to Avaya Communication Manager 2
...
This parameter permits media
encryption between the two Avaya systems joined by this IP trunk
...
This facilitates a key
exchange between the systems, which makes media encryption possible between endpoints on the
two systems, as long as the ip-codec-set forms on both systems are configured with matching
encryption options
...
Media encryption between two systems is possible when they have compatible
codec sets and encryption options, and are connected by an IP trunk with this feature enabled
...

Calls Share IP Signaling Connection: ‘y’ if the far-end is an Avaya device, ‘n’ if it is another vendor’s
device
...
225 signaling connection is used for all trunk members (all calls),
and ‘n’ means that each trunk member (each call) uses a separate signaling connection
...

Bypass if IP Threshold Exceeded: Part of a feature commonly referred to as “TDM fallback” or “IP
trunk bypass
...
The
thresholds for this fail-over are configured in the system-parameters ip-options form, as described
in Appendix G
...

Direct IP-IP Audio Connections: ‘y’ typically, same as with endpoints
...


The LRQ Required parameter allows IP trunk availability to be determined on a per call basis
...
The far-end gatekeeper responds with a RAS-Location Confirm (LCF) message
and the call proceeds
...
If this occurs and the near-end gatekeeper is configured with the necessary route pattern, the
next preferred trunk in the route pattern is used for that call as follows
...

- Wait 2sec for LCF (1sec as of Communication Manager 3
...

- Send LRQ
...
0)
...
0;
2sec total per call as of 3
...

The LRQ feature affects individual calls, whereas the IP trunk bypass feature affects entire IP trunks
...
When this happens, with the appropriate route pattern in place, it results in all calls
being routed onto the next preferred trunk
...

When LRQ is enabled the near-end listen port must be 1719
...
If the far-end gatekeeper is an Avaya call server and also has LRQ
enabled (near-end listen port is 1719), then the near-end gatekeeper must have its far-end listen port set to
1719
...
Each call establishes signaling across the IP trunk after a successful LRQ/LCF exchange
...
323 Signaling and IP Trunk
Groups” at www
...
com
...
This form is used to administer a G700/G350/G250 media
gateway
...
Type is the media gateway model (ie, G700, G350, G250,
G250-BRI)
...
Serial No is the gateway’s serial number, which is displayed by
typing show system at the MGP CLI
...

Network Region is used for IGAR purposes, similar to assigning port networks to a network region on the
cabinet form
...
This is
equivalent to assigning a MedPro/MR320 board to a network region on the ip-interfaces form
...
The default is no automatic recovery (‘none’), or a number can be placed here to apply a
recovery rule, per the system-parameters mg-recovery-rule form, as explained in the following section
...
248 signaling link between the gateway and the call server
...
avaya
...
Site Data can be used to note the gateway’s address (ie, if it is located at a remote
branch office)
...
This is part of the SLS
feature, new to Communication Manager 3
...
The remaining information is automatically
populated when the gateway registers with the call server
...
When a media gateway loses connectivity to the primary call server, it
can fail over to an LSP
...
0, administers rules that determine
when a media gateway automatically recovers back to the primary server
...
Rule Name is a text descriptor
...
248 MG to primary and Minimum time of
network stability are the two conditions that must be met before the primary Communication Manager
server accepts a media gateway recovery registration
...
Then the recovery can happen…
- Immediately
...

- During a specified time window
...

A blank Migrate H
...
The failover to an LSP,
and recovery back to the primary server, are covered in detail in the “Avaya Communication Manager
Network Region Configuration Guide” at www
...
com
...
This form is used for miscellaneous IP settings
...

RTCP Monitor Server: These are the VoIP Monitoring Manager server settings applied to all network
regions, unless specified otherwise in the ip-network-region form
...

- Port network control link between S87xx/S8500 server and IPSI board
...
248 media gateway control link between CLAN/S8300 and media gateway
...


When this feature is enabled, and Communication Manager detects a failure on one of these links,
Communication Manager launches a trace route from the source of the link to the destination of the link
...
This feature should be disabled if ICMP is blocked on
the network, so as not to give false indications
...

H
...
323 IP Endpoint: See the “Avaya Communication Manager Network
Region Configuration Guide” at www
...
com for information on most of the parameters under these
headings
...
This timer determines the frequency at
which a forcefully unregistered IP phone attempts to re-register
...
At some point the user logs off the softphone, leaving the extension free for the IP phone to
reacquire
...
This timer determines
that frequency, and it requires IP telephone 2
...

Music on Hold: This feature applies to media gateways and to port networks in IP-Connect systems with
no traditional PNC (Center Stage or ATM)
...
711 codec for quality reasons
...
711 path between media gateways and port
networks, this feature is not necessary
...
711 path, and in
such cases setting this parameter to ‘y’ forces the use of G
...

IP DTMF Transmission Mode: The intra-system parameter determines how DTMF tones are passed
within a system between media gateways and IP-connected port networks with no traditional PNC
(Center Stage or ATM)
...
Note that both ends of the IP
trunk must be configured the same
...
This is particularly an issue for systems that rely on DTMF tones for
functionality
...

- in-band: If the configured codec is G
...
729, the tones are passed in-band
...
G
...
729
can pass the tones but is susceptible to error
...
1 for intra-system
DTMF digits
...
711, the tones are passed in-band
...
This option removes the uncertainty of G
...
This option
is obsolete on CM 3
...

- out-of-band: The digits represented by the tones are always passed out-of-band
...
245 messages
are exchanged, the H
...
Otherwise, the
Keypad Information Element of an H
...
931 INFO message is used to pass the digits
...
This is required by SIP but also applicable to H
...

The last two options require the MedPro/MR320 board and VoIP media module to detect the tones and
remove them from the outgoing audio stream
...

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Avaya IP Telephony Implementation Guide

44

SAT Troubleshooting Commands
The following table lists some common SAT troubleshooting commands
...

Gives real-time view of a station’s activities – for tracing calls
...


status signaling-group
status trunk or
/

Gives status of signaling-group
...


status ip-board
status clan-port (ie, 01a0517)
status clan-usage
status media-processor all | board

Gives Ethernet interface in/out statistics for an IP board
...

Gives C-LAN socket usage
...


status ip-network-region <#>

Gives status of inter-region connectivity
...

Gives administered vs
...

Sends pings and trace-route from a board or from a station
...
If station, specify source ,
where port # is from status station form
...


Table 7: Common SAT troubleshooting commands

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4

Guidelines for Avaya 4600 Series IP Telephones

This section covers some general information regarding various Avaya 4600 Series IP Telephone models
...
avaya
...
The current GA firmware releases can be obtained at the
same site
...

Note: For simplicity in many IP telephone applications a C-LAN is often called a gatekeeper, although
the call server is the gatekeeper and the C-LAN is only a front end to the gatekeeper
...
1

Basics

Legacy Models vs
...
The last and
best firmware for these models is version 1
...
3; legacy models cannot accept firmware newer than this
...
When
connected to an Ethernet switch port that is configured to auto-negotiate, the Ethernet switch port
stabilizes at 100/half
...
In this case, all three devices stabilize at 10/half
...
If a personal computer might be attached to
the telephone, and there is a chance that the computer might have a 10-mbps NIC, leave the Ethernet
switch port in auto-negotiate mode
...

The term “current models” in this document refers to the 4620 and 4610 and models containing the SW
designation
...
Thus, SW models do not need, or work with, the
30A (applicable to the 4612/4624/4630 only) switched hub interface
...
The built-in 10/100 Ethernet switch
permits speed and duplex configuration if necessary
...
The closest Ethernet switch to which the IP Telephone is attached should be set to
auto-negotiate, as well
...
Follow the
guidelines in section 2
...
Current models also have an updated
look and a larger screen that facilitates additional features and functionality
...
avaya
...

When the IP phone and PC are both transmitting, the phone’s traffic is given strict priority out the uplink
port to the enterprise Ethernet switch
...
Prioritization of traffic downstream from the
enterprise Ethernet switch to the phone’s switch port must be handled by the enterprise Ethernet switch
...
1Q tag from the IP telephone toward the PC
...
This also allows the attached PC to communicate with the IP telephone when they
are on the same VLAN and the phone is tagging
...
See the “4600 Series IP Telephone LAN
Administrator’s Guide” for more details
...
Each option is
associated with a specific bit of information to be sent by the DHCP server to the DHCP client
...
Option 3 is
the router option and is used to send the default gateway address and other gateway addresses to the
client
...
The defined options are
found in RFC 2132
...
They are standard options that are not defined, and
vendors may use these options and define them to be whatever is necessary for a specific application
...

For the Avaya application of option 176, it is defined as a string
...
The most prevalent parameters and
values are as follows
...
1Q VLAN ID – 0 default
L2QAUD
L2 audio priority value
...

VLANTEST
The number of seconds a phone will attempt to return to the
previously known voice VLAN
Table 8: DHCP option 176 parameters and values

The typical option 176 string for a single-VLAN environment looks like this
...
MCPORT specifies which UDP port to use for RAS
registration
...
6
...
A TFTP server address is necessary so that phones know where to go to download the
necessary script files and binary codes (see “Boot-up Sequence” heading below)
...
1Q tagging were required, such as in a dual-VLAN environment
(see section 4
...
Other parameters may be added, such as L2QAUD and L2QSIG, which are used to
specify the L2 priority values for audio and signaling
...

Note: The L3 priority values (DSCP) are received from the call server, as configured on the SAT ipnetwork-region form
...
The preferred and recommended method is
via DHCP option 176
...
5, heading “ip-network-map,”
which utilizes the L2 values administered on the SAT ip-network-region form
...
Option 176 could be applied globally or on a per scope basis
...
As part of the DHCP process at boot-up, the IP
telephone requests option 176 from the DHCP server
...
The DHCP specification calls for the client to renew the lease at
determined intervals, typically beginning at half-life of the lease
...
Too
short a lease requires too many renewals, which not only taxes the DHCP server but can also disrupt
service to the IP phones if renewals cannot be accomplished for whatever reason
...

Additional Script and Firmware Download Methods
Beginning with Avaya IP Telephone Release 2
...
Preliminary testing at
Avaya Labs indicates that HTTP servers can support more simultaneous downloads than TFTP servers,
suggesting that HTTP/TLS are better suited for large IP telephone deployments than TFTP
...
If TLSSRVR,
HTTPSRVR, and TFTPSRVR are all set, the phone will attempt to download firmware using TLS first on
TCP port 411, then HTTP on TCP port 81, then HTTP on TCP port 80, then TFTP on UDP port 69
...

Boot-up Sequence
The following are key boot-up events, listed in order, which may help to verify proper operation of the IP
phone
...
The packets described here can
be captured using a protocol analyzer, and one with H
...
225 RAS messages
...
But
because the 4620 and 4610 have a built-in switch instead of a hub, the analyzer must be attached to a
mirrored Ethernet switch port, or to a tap or hub in line between the phone and the Ethernet switch
...

The display shows Restarting… (if the phone was intentionally restarted w/ Hold RESET#), and
then Loading… and Starting…
- DHCP – The phone queries the DHCP server for an IP address and other needed information
...
Note that this step is bypassed if the phone is manually
configured with all the necessary information
...
scr” and others from TLS/HTTP/TFTP server – This is a text script file
that tells the phone which boot code and application code are needed
...
A brand new phone makes all three requests, as
phones typically come from the factory with outdated code
...
scr” script
may instruct the phone to download the “46xxsettings
...
txt)” file, which is an optional method
of sending configurations to the phone
...
bin code is
received for the first time
...

- Ext and Password prompts – The phone prompts for the extension and password if there are no
previously stored values
...
This registration happens very quickly and does not show up on
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Avaya IP Telephony Implementation Guide

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-

-

the display
...

H
...
225 Setup message, which is answered with H
...
225 Connect messages
from the gatekeeper
...
During idle
periods the phone maintains the session by sending TCP keepalives
...

Unregistration messages – If the gatekeeper intentionally unregisters a set, or if the set intentionally
unregisters itself, the message sent by either the gatekeeper or the set is a RAS-Unregistration
Request (URQ) with a reason code that is deciphered in the hex decode of most protocol analyzers
...


Call Sequence
It is not feasible to give a standard packet-by-packet call sequence, because of the many possible
variations on any given call
...
Depending
on which features are enabled and executed during a call the packet-by-packet sequence may vary, but the
fundamental functions described here apply overall
...

- Calling phone contacts gatekeeper on already established call signaling session (TCP 1720 gatekeeper
port, variable phone port)
...

- Calling phone establishes an audio stream with an audio resource (MedPro/MR320 board or VoIP
module), as directed by the gatekeeper
...

- There are some call signaling exchanges on this TCP session
...

- Called phone answers, resulting in more call signaling activity, and the call completes
...

- Phones direct audio streams to each other, as instructed by the gatekeeper
...

- Gatekeeper contacts both phones, signals that the call has ended, and instructs them to tear down
audio streams
...

Keepalive Mechanisms
There are two types of keepalive mechanisms: RAS and TCP
...
On a protocol analyzer a RAS keepalive message shows
up as a RAS-Registration Request (RRQ) with the keepalive bit set in the RAS decode
...
This exchange
takes place over the RAS socket, which has UDP port 1719 on the gatekeeper side
...
The keepalive
is an empty TCP datagram with a sequence number that is 1 to 5 less than the sequence number of the
previous real TCP message or ACK sent by the phone
...
This exchange takes place over the call
signaling socket, which has TCP port 1720 on the gatekeeper side
...
However, because a CLAN must keep track of potentially
hundreds of phones, the CLAN’s keepalive intervals are much longer than the phone’s keepalive
intervals
...
These keepalives are
not synchronized, so they don’t all go out to every phone at the same time
...
So a link bounce
detection time for a CLAN is 5-15min
...
This means it takes CM 5-15min to internally unregister that phone
...

Regular and retry intervals – Each keepalive mechanism has a regular interval as described above
...
If all the retry keepalives are unanswered, the phone effectively unregisters and moves on
to the next gatekeeper in its gatekeeper list (obtained via DHCP and/or the gatekeeper)
...
The TTL is
the greater of 60 seconds or a multiplier times the number of registered endpoints
...
4 seconds, which means that anything above 42 registered endpoints
would exceed the minimum 60-sec TTL
...
1 second, which means that more than 600 registered endpoints are required to exceed
the minimum 60-sec TTL
...

regular interval

retry int

retry int

retry int

retry int

retry int

retry int

KA
retry KA retry KA retry KA retry KA retry KA
no ACK no ACK no ACK no ACK no ACK no ACK
time to unregister

discovering

KA
ACK

failure

KA
ACK

regular interval

Figure 17: Keepalive pattern

The discovering at the end of the flow means that the phone has effectively unregistered and is searching
for another gatekeeper
...
Even if the phone did send a URQ, chances are the gatekeeper would not receive
it because the failure condition could still exist
...

And indeed if the phone did receive a KA acknowledgment within that final retry interval it would stay
registered to the same gatekeeper
...
Therefore, the
final retry interval really does not factor into the time to unregister
...
If the failure recovers a couple seconds after the final retry KA is sent, the phone most likely
unregisters and moves on to the next gatekeeper after the final retry interval
...

IP telephone
4620/10
2
...
x and later

TCP KA
regular intrvl
20sec
configurable

TCP KA
retry intrvl
5 * 5sec
configurable

Time to
unregister
25 to 45sec
varies

RAS KA
regular intrvl
obsolete

RAS KA
retry intrvl
Obsolete

Time to
unregister
n/a

Table 9: TCP and RAS keepalive matrix
4
...
(The port with
the icon that looks like a network jack is the uplink port, which connects to the Ethernet switch
...
For example, do not connect an enterprise server to the phone
...
Also, do not
connect a PC to the phone with a 10M uplink to the network
...

IP Phone and Attached PC on Same VLAN
There are three variations of attaching a PC to the phone, and the first two involve having both the phone
and the PC on the same VLAN, which is the port/native VLAN (refer to Appendix A for a primer on
VLANs)
...
In this case,
no special configurations are necessary
...

The second scenario is similar to the first, except that traffic from the phone is marked with L2 and/or L3
priority while remaining on the port/native VLAN
...
3 under the heading
“Rules for 802
...
” The phone must be configured to apply the appropriate L2 and/or L3
priority values
...
1Q tagging and to set
the VLAN ID
...
The manual method is
covered below, and an automated method is covered in the next paragraph
...
1Q – On/off for 802
...
Turn this on if L2 priority tagging is desired; off otherwise
...
3, heading “Rules
for 802
...
” The VID has no effect when 802
...

- VLANTEST – Not relevant when VID is zero
...

- L2 audio – Layer 2 CoS tag for Ethernet frames containing audio packets
...
This
value could also be set manually on a per phone basis
...
The phone either
receives this from DHCP (most common) or from the call server (rare), per the ip-network-region
form
...

- L3 audio – Layer 3 DSCP for audio IP packets
...
This value could also be set manually on a per phone
basis
...
The phone automatically receives this value
from the call server, per the ip-network-region form
...

The manual menus are covered here for explanatory purposes
...
As stated previously, the call server sends the L3 priority values to the phones
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automatically, per the values configured in the ip-network-region form
...
1Q on/off instruction,
VLAN ID, and L2 priorities can be configured automatically using DHCP option 176 as described in
section 4
...
” Here is what that string should look like for 1
...

MCIPADD=addr1,addr2, … , ,HTTPSRVR=addr,L2QVLAN=0,L2QAUD=#,L2QSIG=#
The L2QVLAN=0 parameter instructs the phone to enable 802
...
The Ethernet switch port to which the phone is
connected must be configured to accept 802
...
1Q standard [6 p
...
If the Ethernet switch does not
understand VID 0, the phone may need to tag with the port/native VID, although this is not the standard
method
...
Remember also that improperly enabling L2 and
L3 prioritization may break processes that were working without it
...
3 of this document for
more information on CoS and QoS
...
This requires a dual-VLAN port on
the Ethernet switch as described in section 2
...
1p/Q Tagging
...
1Q tag)
from the PC are forwarded on this VLAN
...

The Hold ADDR# and Hold QOS# menu options are the same as described in the previous heading,
except that the VID must not be zero
...
1,
heading “DHCP Option 176”) is also the same, except that L2QVLAN has a non-zero value
...

The following scenario, with arbitrary voice VLAN ID, details the steps a phone (1
...
It also illustrates the recommended content of the
option 176 string
...

- The data VLAN option 176 string directs the phone to go to voice VLAN 25
...


-

The voice VLAN option 176 string is identical to the data VLAN string but without the L2QVLAN
parameter, because a phone already on the voice VLAN doesn’t need to be directed to go there
...

Reboot or power cycle occurs
...

- Phone obtains an address and option 176 string on the voice VLAN and all is well
...
In this
case the VLANTEST=600 parameter directs the phone to continue trying for 600sec (finite range
is 1-999)
...


MCIPADD=addr1,addr2, … ,MCPORT=1719,TFTPSRVR=addr,L2QAUD=6,L2QSIG=6,VLANTEST=600

The idea behind going back to the data VLAN after some time is that the phone may have changed ports
and be on one with a different voice VLAN
...
The idea behind marking VLAN 25 as invalid in the previous
scenario is that if the phone hasn’t changed ports, it is preferable to operate on the data VLAN than to be
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sent to a bad voice VLAN in a continuous loop
...
8
...
0
...
This instructs the phone to permanently remain on the previously known voice VLAN
...

Note: DHCP option 176 is the preferred method for directing IP phones to the voice VLAN
...
The two methods should not be used simultaneously
...
4
...
The value range increases from 0

- 999 (16
...
Also, when the timer expires, the voice VLAN is
NOT marked as invalid
...
If it is not answered it will shift back to the voice vlan for the same value of VLANTEST
seconds
...
You no longer have
to manually reset values to clear the information from memory
...
It will disable moving from the voice VLAN back to the data VLAN by keeping all requests on the
voice VLAN
...

Remember that in order for the CoS markings to have any effect, the corresponding QoS configurations
must be implemented on the necessary network devices
...
Read section 2
...

4
...
This list is obtained via the DHCP option
176 string, which is covered briefly in section 4
...
”
Within the DHCP option 176 string, the comma-separated IP addresses that follow the MCIPADD
parameter constitute a gatekeeper list, and this list provides redundancy at boot-up
...
The following
hypothetical network diagram and the accompanying instructions explain how gatekeeper lists should be
administered on DHCP servers
...
The main site is implemented in a core-distribution-access architecture common to many
enterprise networks
...
Network region 1 has
four C-LANs scattered across four distribution switches, but there could be more depending on the
number of IP telephones
...

Suppose, for whatever reason, that a large number of IP phones are rebooted at once
...
All the phones should not bombard the same gatekeeper at once with GRQs
...

- v10 scope: “MCIPADD=clan1addr,clan2addr,clan3addr,clan4addr, …”
- v20 scope: “MCIPADD=clan2addr,clan3addr,clan4addr,clan1addr, …”
- v30 scope: “MCIPADD=clan3addr,clan4addr,clan1addr,clan2addr, …”
- v40 scope: “MCIPADD=clan4addr,clan1addr,clan2addr,clan3addr, …”
Based on how this particular network is implemented, here is another alternative
...
DHCP scopes should have
rotating/varying gatekeeper lists, so as to produce a uniform distribution of GRQs at boot-up
...
Do not create a global option 176 string that would apply to every scope on a server,
resulting in only one gatekeeper list
...

Branch Site
The branch site is just slightly different in terms of the DHCP scopes, but very different in terms of the
failure scenario and other factors that affect the branch implementation
...
In either case the DHCP scopes for v80
and v90 should have rotating lists, as at the main site
...
This
is because the LSP can take over as the call server for the branch if the WAN link fails
...

Because an extended WAN link failure is possible, the branch site should ideally have its own DHCP
server
...
For cost and administrative reasons,
however, many will choose not to install a DHCP server at all branch locations
...
The manual configuration option is available, but it is not always a viable
option for various reasons
...
In other words, when an IP telephone registers
with the call server, the call server sends a gatekeeper list in the RCF message
...
323 standard calls
this the Alternate Gatekeeper List
...
This feature is useful for phones
that are manually administered, as the manual method only permits the entry of one gatekeeper address
...

Here are some key points regarding the option 176 gatekeeper list and the RCF Alternate Gatekeeper List
...
0 use both lists simultaneously
...

- IP telephone 2
...

During boot-up the phone uses the list obtained from option 176
...
When a phone is logged off but not rebooted, it
reverts back to the list obtained from option 176
...
If the phone only knows of one GK at bootup and that GK is out of service, the phone cannot register and hence cannot get an RCF
...
When an IP phone
registers and its network region is specified in the ip-network-map form, the call server delivers a
list of all gatekeepers in that region, plus directly connected regions (specified in the ip-networkregion form)
...
The ip-interface form includes an administered Gatekeeper
Priority Value between 1 and 9 where 1 is the highest priority
...
If multiple gatekeepers have the same
priority value then the gatekeeper list is based on socket load per gatekeeper within the same priority
...
3, the addresses of the LSPs (administered on the ipnetwork-region form) in the same network region as the IP phone are also sent in the RCF
...
0, in addition to the LSPs, the address of the Survivable GK Node Name
(administered on the station form) is also sent in the RCF
...
x and IP telephone 2
...
During recovery after
an outage, the primary gatekeepers are attempted first for a period of time called the H
...
After this search time expires,
the secondary gatekeepers – LSPs and the Survivable GK – are also included in the search
...
323 Link Bounce section of the “Avaya Communication Manager
Network Region Configuration Guide” at www
...
com
...

Phone logged off via Hold
Phone state Registered phone
LOGOFF# keypad command
Method
- 2
...
1 and later shows
list received from option 176, or
Alternate Gatekeeper List
manually configured
received in RCF message,
gatekeeper
...

- 2
...
1 shows gatekeeper list
- 2
...
1 shows gatekeeper list
received from option 176, or
MIB Object ID
received from option 176,
manually configured

...
3
...
1
...
1
...
2
...
1
...
3
or manually configured
gatekeeper
...

- 1
...
x shows combined list from
RCF and option 176, or
- 1
...
x shows combined list
combined RCF list and
from RCF and option 176,
manually configured
or combined RCF list and
gatekeeper
...

Shows gatekeeper to which
Shows gatekeeper to which phone
MIB Object ID
phone is currently registered
...


...
3
...
1
...
1
...
2
...
1
...
4
(endptMCIPINUSE)
2
...
2 and later shows the alternate
MIB Object ID
alternate gatekeeper list
gatekeeper list received from CM in

...
3
...
1
...
1
...
2
...
1
...
4
...


KW

Avaya IP Telephony Implementation Guide

56

Appendix A: VLAN Primer
This appendix is primarily concerned with configurations that require the Avaya IP Telephone to
connect to an Ethernet switch (Eth-switch) port configured with multiple VLANs – the IP phone on one
VLAN and a PC connected to the phone on a separate VLAN
...
2
...

VLAN Defined

With simple Eth-switches, the entire switch is one L2 broadcast domain that typically contains one IP
subnet (L3 broadcast domain)
...
A VLAN is a logical L2 broadcast domain that typically contains one
IP subnet
...
A L3 routing process is required to route between VLANs, just as
one is required to route between switches
...
If there is no routing process associated with a VLAN, devices
on that VLAN can only communicate with other devices on the same VLAN
...
avaya
...

The Port or Native VLAN

Port VLAN and native VLAN are synonymous terms
...
11], but Cisco switches use the term native VLAN
...

Every port has a port/native VLAN
...
It can be
configured on a per port basis with the following commands
...
1Q tag, such as from a PC) are forwarded on the
port/native VLAN
...
1Q trunk, or otherwise
configured for multiple VLANs (see VLAN binding heading below)
...
IEEE 802
...

Cisco also uses a proprietary method called ISL
...

A trunk link is a connection between two devices across trunk ports
...
Some form of trunking or
forwarding multiple VLANs must be enabled to permit the IP phone and the attached PC to be on
separate VLANs
...

Avaya P330 and C360

Cisco CatOS

set trunk dot1q

set trunk nonegotiate dot1q

By default only the port/native VLAN is enabled on
the trunk port
...


By default all VLANs (1-1005) are enabled on the
trunk port
...


Note that Avaya adds additional VLANs to a trunk port that has only one VLAN, while Cisco removes
excess VLANs from a trunk port that has all VLANs
...

VLAN Binding Feature (P330/C360)

On the Avaya P330/C360, additional VLANs are added to a port using the VLAN binding feature
...
1Q tagging enabled) or an access port (no 802
...
The port does
not need to be a trunk to forward multiple VLANs, and for one application – connecting to an Avaya IP
phone – it must not be a trunk (ie, do not issue the set trunk command)
...

1
...

2
...

Static option:
set port vlan-binding-mode static
set port static-vlan

Put the port in bind-to-static mode
...


----- OR ----Configured option:
set vlan
set port vlan-binding-mode bind-toconfigured
3
...
Type show
vlan to see entire list
...


If the port is connected to a router or to another switch, trunking must be enabled with the command set trunk
dot1q, which causes all egress frames to be tagged
...
This
is necessary because most PCs do not understand tagged frames
...
That is, the Avaya switch does not need to be explicitly
configured to accept priority-tagged Ethernet frames on a port with only the port/native VLAN
configured
...
Simply enable 802
...
Per the IEEE standard, a VID of zero assigns the
Ethernet frame to the port/native VLAN
...
Here are Avaya Labs test results with a sample of hardware platforms and OS versions
...
1(2)

Catalyst 4000 w/
CatOS 6
...
0(5)WC2
Conclusion

KW

Accepted VID zero for the native VLAN when 802
...
In this case, all but the native VLAN should be cleared
off the trunk
...
Opened a case with
Cisco TAC, and TAC engineer said it was a hardware problem in the 4000
...
Workaround is to enable 802
...
Again, clear all but the native VLAN off
the trunk
...
1Q trunking was
disabled on the port
...

Avaya IP Telephony Implementation Guide

58

Note that setting a L2 priority is only useful if QoS is enabled on the Eth-switch
...

Sample Multi-VLAN Scenario for Avaya P330 Code 3
...
8 and Cisco CatOS and IOS

Here is a sample multi-VLAN scenario
...
To conserve ports and cabling,
the PCs are connected to the phones and the phones are connected to the P330 switch
...
168
...
1

MedPro
vlan 10
192
...
10
...
168
...
254
192
...
10
...
168
...
7

1/5

Avaya
IP Phone

Cajun P330
1/12
vlan 1
192
...
1
...
168
...
7

DHCP Server
TFTP Server

PC

Cisco Router configuration
interface FastEthernet0/1
description 802
...
1
encapsulation dot1q 1
ip address 192
...
1
...
255
...
0
!
interface FastEthernet0/1
...
168
...
254 255
...
255
...
168
...
100

To forward DHCP requests to the DHCP server
...

Port in static binding mode by default, but command shown
...

set port static-vlan 1/1 10
Port connected to Cisco router is an 802
...

set trunk 1/1 dot1q
Spanning Tree disabled at the port level
...

Spanning Tree disabled at the port level
...
1p) priority set to 6
...

In addition to v1, v10 statically bound to port, but not a trunk port
...

Port 1/12 for the DHCP/TFTP server already has port/native VLAN 1
...


P330/C360 configuration (bind-to-configured option)
All ports have port/native VLAN 1 by default
...


set port vlan 10 ½
set port spantree disable ½
set port level ½ 6

Port/native VLAN changed to 10 on this port
...

Port L2 (802
...


Port connected to Cisco router is an 802
...

Spanning Tree disabled at the port level
...

Spanning Tree disabled at the port level
...

First invoke this command on all user ports
...

set vlan 1005 1/1
Port connected to Cisco router is an 802
...

set trunk 1/1 on dot1q
Unnecessary VLANs removed; 1, 10, and 1005 remain
...

Port L2 (802
...


set vlan 10 ½
set port qos ½ cos 6
set vlan 10 1/3
set port qos 1/3 cos 6

set port auxiliaryvlan 1/5 10

set trunk 1/5 nonegotiate dot1q
clear trunk 1/5 2-9, 11-1005

Auxiliaryvlan is the more common method instead of explicit
trunking
...
1Q trunk port, though not explicitly configured
...

Plain 802
...

Unnecessary VLANs removed; 1 and 10 remain
...

interface FastEthernet0/1
KW

Avaya IP Telephony Implementation Guide

60

switchport trunk allowed vlan 1,10,1005
switchport mode trunk
spanning-tree portfast

Port connected to Cisco router is an 802
...

Cisco switches do not tag the native VLAN, but the router expects a
tag on v1, so the native VLAN is changed to some unused VLAN
...

Port is in trunk mode
...


interface FastEthernet0/2
switchport access vlan 10
spanning-tree portfast
switchport priority default 6

Port/native VLAN changed to 10 on this port
...

Port native VLAN L2 (802
...


switchport trunk encapsulation dot1q
switchport trunk native vlan 1005

interface FastEthernet0/3
switchport access vlan 10
spanning-tree portfast
switchport priority default 6
interface FastEthernet0/5
switchport trunk encapsulation dot1q
switchport trunk native vlan 1
switchport trunk allowed vlan 1,10
switchport mode trunk
spanning-tree portfast

802
...

Since most PCs do not understand the tag, the PC’s VLAN must be
the native VLAN
...

VLANs 1 and 10 allowed on trunk
...

Spanning Tree fast start feature
...

Access mode; explicit trunking not required
...

Configure the voice VLAN
...
Initially placing the IP phone on VLAN 10
requires two DHCP scopes – one for VLAN 1 and another for VLAN 10
...
The VLAN 1 scope must have the L2QVLAN
parameter, and the VLAN 10 scope should not
...
8 and
beyond
...
It obtains an IP address on VLAN 1 – the
port/native VLAN
...
After the phone is operational on VLAN 10, on subsequent reboots the phone returns
to VLAN 10 directly, without passing through VLAN 1
...
See section 4
...

The L2QVLAN parameter should not be added to the VLAN 10 DHCP scope
...
In such a case the phone would not require tagging
to function on VLAN 10, and tagging could result in an incompatibility with the Eth-switch
...
No special configurations are required
...
Substantial testing and production operation have
shown that Avaya IP phones interoperate with both auxiliaryvlan (CatOS) and voice vlan (IOS), and
these have become the preferred methods of implementation over explicit 802
...
This
interoperability research was initiated because of the inability to enable portfast on older Catalyst 6500
code (pre 5
...
14, 6
...
2, 7
...
2) when the port is in trunk mode
...

Interoperability with auxiliaryvlan and voice vlan was successfully lab tested on the following
platforms, with no known issues to date
...
2
...
3
...
5
...
5
...
5
...
2
...
3
...
5
...
5
...
0(5)WC11
voice vlan on Catalyst 3550 with IOS version 12
...
1x
Furthermore, Avaya IP phones have been deployed on a broader range of CatOS and IOS platforms by
various Avaya customers, also with no known issues to date
...
1Q trunking are all viable options when a dualVLAN environment is required (see Appendix A)
...
802
...

For IOS-based Catalyst switches, voice vlan is roughly equivalent to auxiliaryvlan
...
On
newer IOS platforms (ie, 3550, 3560), however, voice vlan can be enabled without explicit 802
...

Note that Avaya IP phones do not interoperate with CDP
...
The Avaya IP
phone can learn the auxiliaryvlan/voice vlan designation via DHCP option 176, as explained below and in
Appendix A
...

At the heart of Cisco’s auto-discovery feature are Cisco-proprietary mechanisms
...
This is a layer 2 protocol, which means that it works at
the Ethernet level, without requiring IP addresses
...
(CDP packets can be
captured and decoded using protocol analyzers that support CDP
...
[1 p
...
The auxiliaryvlan is the second Ciscoproprietary mechanism, and it must be enabled on the port that connects to the IP phone
...
According to Cisco’s
documentation the auxiliaryvlan is just another 802
...
The only difference is the proprietary
method of assigning it to a Cisco IP phone
...
This implies that the port is an 802
...
1p/Q tagged frames
...
2
...
[1 p
...
The phone
communicates its specific power requirements to the Catalyst, and the phone can also trigger the Catalyst
to send its CDP packet immediately instead of waiting for the transmit period (60 seconds by default) to
recycle
...
2-23]
Avaya IP Phones on Cisco Auxiliaryvlan

The auxiliaryvlan is a modified method of implementing 802
...
Although testing to date has been positive, Avaya does not know what other mechanisms are or
will be incorporated with this feature, or if they could have any adverse effects on Avaya IP phones
...
1Q trunk port, the following steps
allow Avaya IP phones to work on Cisco’s auxiliaryvlan
...

a) For example, the command set port auxiliaryvlan 2/4-8 500 would make ports 2/4 through 2/8
auxiliaryvlan-capable with auxiliaryvlan ID 500
...

c) The command show port auxiliaryvlan reveals the ports that have been made auxiliaryvlancapable, and their respective auxiliaryvlan ID(s)
...

2) Bring up the phones on the auxiliaryvlan using the same procedures that would be used on a regular
trunk port
...
Both interfaces must be configured to forward
DHCP requests (ip helper-address ...

b) Follow the instructions at the end of appendix A to get the IP phone on the auxiliaryvlan (voice
VLAN)
...

3) For call servers, IP boards (ie, C-LAN and MedPro/MR320), and other VoIP resources, configure
their ports on the Eth-switch to be native to the auxiliaryvlan
...
Just make the auxiliaryvlan the port/native VLAN on these
ports (set vlan 200 , assuming 200 is the auxiliaryvlan ID)
...

4) Always verify network connectivity between devices using pings and trace-routes
...
It is a function of the network and not a function of the VoIP application
...

Application Perspective

Here is the anatomy of a 20-ms G
...
Notice that two-thirds of the packet is consumed by overhead (IP, UDP, and
RTP), and only one-third is used by the actual audio
...
729 Audio
20B

It is important to understand that all 20-ms G
...
Not only is the structure of the packet the same, but the method of encoding and decoding the
audio itself is also the same
...
The packets from the
application perspective are identical
...
1] [2 p
...
Cisco routers employ this mechanism, as does
the Avaya X330WAN router, which is a module for the P330 chassis
...
729 audio
...
This equates to reducing the total VoIP WAN bandwidth consumption by roughly
half, and it applies to all 20-ms G
...

Customers who deploy routers capable of this feature may be able to benefit from it
...
Depending on the processor load before compression, enabling RTP
header compression could significantly slow down or crash the router
...

RTP header compression has to function with exactness or it will disrupt audio
...
This has been very difficult to quantify, but
there is some anecdotal evidence
...
When RTP header compression was
disabled, simply for experimentation purposes, the audio problems went away
...
Although this test was conducted using Cisco routers, the expected behavior is the same for
any router that performs this function as specified in RFC 2508 [7]
...


KW

Avaya IP Telephony Implementation Guide

65

V
...
0 was used to simulate VoIP calls between the two endpoints
...
0
accurately simulates the characteristics of various codecs and uses a 40-byte IP/UDP/RTP header
...
50
...

The Cisco 3600 had IOS v12
...
0(12)
...
35 serial link to take
bandwidth measurements
...


-

A single call was placed between the Chariot endpoints using the two most common codecs, sending
20-ms voice packets
...
Note that these
are rough measurements
...
711 (64 kbps)
G
...
5
27
...
This was
done by spot-checking the audio packets before and after compression, using two Sniffer protocol
analyzers
...
729 the RTP header and payload were identical before and after compression
...
711, however, the received packets had the PADDING flag set in the RTP header, although the flag
was not set when the packets were transmitted
...
711
...
711 codec if bandwidth is scarce
...
Specify the number of RTP connections that can be compressed (cache allocation)
...
The default is 32,
and each call requires two connections
...
3 and later; and 3 to 1000 for PPP and HDLC using IOS v12
...
For Frame
Relay the value is fixed at 256
...
The command to turn on compression is ip rtp header-compression in interface configuration mode
...
For this experiment, when the command was
entered into the router, ip tcp header-compression was also installed automatically
...

Consult Cisco’s documentation for more specific configurations on other types of WAN links (ie, Frame
Relay and ATM) [2 p
...
Configuration for the X330WAN router is very similar
to Cisco and well documented in the X330WAN User Guides
...
The ports used by the Avaya call server are fairly fixed and known
...
As a result, it is simpler to tailor access lists based on call
server ports
...

Permit Any C-LAN
UDP 1719
Any endpoint
UDP any
Permit Any endpoint
UDP any
Any C-LAN
UDP 1719
The C-LAN uses TCP port 1720 for H
...

Permit Any C-LAN
TCP 1720
Any endpoint
TCP any
Permit Any endpoint
TCP any
Any C-LAN
TCP 1720
This is to facilitate IP trunking between two Avaya call servers, and must be done for each IP trunk
...

Permit Any MedPro/MR320
UDP port range
Any endpoint
UDP any
in ip-networkregion form
Permit Any endpoint
UDP any
Any MedPro/MR320
UDP port range in
ip-network-region
form
This is another way to facilitate audio streams between MedPros/MR320s and endpoints
...

Permit Any endpoint
Any endpoint
UDP any
UDP any
RTP/RTCP
-The R300 uses this default UDP port range for audio
...

Permit Any R300
Any MedPro/MR320 or
UDP 1900-2075
UDP varies
endpoint
RTP/RTCP
-Permit Any MedPro/MR320 or
Any R300
UDP varies
UDP 1900-2075
endpoint
RTP/RTCP
-Permit Any R300
Any R300
UDP 1900-2075
UDP 1900-2075
RTP/RTCP
-These are all services used by the IP telephone
...
The GET and PUT
requests from the client go to the server’s UDP port 69, but all other messages go between random ports
...
For example, C-LANs ping endpoints for management
purposes; MedPros/MR320s ping C-LANs to gauge network performance across an IP trunk; IP telephones ping
TFTP servers for verification purposes
...
Most
connections take place over the S8xxx server’s enterprise interface, which could be a separate interface or
combined with a control network interface
...

Typically eth0 on S87xx IP-Connect, but could also be configured as eth4 in some cases
...


Action

TCP/UDP port
To
TCP/UDP port
or Protocol
or Protocol
This allows the Communication Manager 1
...
A
TCP session is initiated from the primary server to the LSP TCP port 514
...

Permit Primary server enterprise intfc TCP any
LSP
TCP 514
Permit LSP
TCP 514
Primary server enterprise intfc TCP any
Permit LSP
TCP any
Primary server enterprise intfc TCP 512-1023
Permit Primary server enterprise intfc TCP 512-1023
LSP
TCP any
This allows the Communication Manager 2
...

Permit Primary server enterprise intfc TCP any
LSP
TCP 21873
Permit LSP
TCP 21873
Primary server enterprise intfc TCP any
This allows the Communication Manager 3
...

Permit Primary server enterprise intfc TCP any
LSP
TCP 21874
Permit LSP
TCP 21874
Primary server enterprise intfc TCP any
This allows an administrator to log in via Avaya SA to a call server (S87xx, S8500, S8300)
...
The call server redirects
unsecure sessions to https
...

Permit S8xxx enterprise interface
UDP any
DNS server(s)
UDP 53 (dns)
Permit DNS server(s)
UDP 53 (dns)
S8xxx enterprise interface
UDP any
Permit S8xxx enterprise interface
UDP any
NTP server(s)
UDP 123 (ntp)
Permit NTP server(s)
UDP 123 (ntp)
S8xxx enterprise interface
UDP any
H
...
MG initiates session
...
248 encrypted signaling between G700/G350/G250 Media Gateway and S8300 or CLAN
...

Permit G700/G350/G250
TCP any
S8300 or CLAN
TCP 1039
Permit S8300 or CLAN
TCP 1039
G700/G350/G250
TCP any
Control network traffic and other traffic between S87xx/S8500 and IPSI board
...

sets the port speed
sets the port duplex
enables Spanning Tree fast start feature (no to undo)
sets the native vlan (default vlan) when port is in access mode
(default is access mode, where there is only one vlan on port)
puts port in trunk mode
makes trunk 802
...
actual speed and duplex for an IP board

change ip-interface
list ethernet-options

get ethernet-options
IPSI commands
set port negotiation 1 enable|disable
set port speed 1 100MB|10MB
set port duplex 1 full|half
show port 1
show control stats

KW

These commands are executed from the IPSI [IPADMIN] prompt
...
It is only meant to give the
reader a starting point
...

This rudimentary network configuration is used as a reference point
...

other subnet

C-LAN

other subnet

C-LAN
MedPro
MedPro

Ethernet0

Ethernet0

Router

Serial0

WAN

Serial0

Router

192
...
1
...
168
...
0/24
other subnet

other subnet

Example 1 – Ideal / WAN terminating on G700/G350/G250 gateway
Suppose all endpoints are capable of marking with one DSCP for audio and another DSCP for signaling
...
Previous firmware versions and the TN799C board cannot mark at L2 or L3
...

create a class map called voipAudio
class-map match-any voipAudio
any packet with DSCP 46 is in this class
match ip dscp 46
class-map match-any voipSig
match ip dscp 34
class-map match-any ipsiSig
match ip dscp 36

create a class map called voipSig
any packet with DSCP 34 is in this class
create a class map called ipsiSig
any packet with DSCP 36 ( af42 ) is also in this class

policy-map voipQoS
class ipsiSig
bandwidth 128
class voipAudio
priority 768
class voipSig
bandwidth 48

create a policy map called voipQoS
give packets in the ipsiSig class 128K
of this WAN link
reserve 768k of this WAN link for packets in the voipAudio class

class class-default
fair-queue
random-detect dscp-based
interface Serial0
description T1
ip address 172
...
0
...
323 and IPSI signaling are put in separate queues and
receive different treatment
...
When terminating a WAN link to these devices, audio must be marked
with DSCP 46, and signaling with DSCP 34 and IPSI signaling with 36
...

create a class map called VoIP
class-map match-any VoIP
any packet with DSCP 46 is in this class
match ip dscp 46
create a class map called IPSI
class-map match-any IPSI
any packet with DSCP 36 (af42) is in this class
match ip dscp 36
create a policy map called voipQoS
policy-map voipQoS
guarantee bandwidth to IPSI traffic
class IPSI
prioritize packets in the VoIP class and dedicate 816k
bandwidth 128
of this WAN link
class VoIP
put everything else in the default class and transmit it out the default
priority 816
queue in a weighted fair queue fashion
class class-default
if the default queue starts to get full, randomly discard packets in this
fair-queue
queue based on DSCP (lower values get discarded first)
random-detect dscp-based
interface Serial0
description T1
apply the voipQoS policy outbound on this interface
ip address 172
...
0
...

Separate Queues vs
...
When considering the three detriments to
IP telephony – delay, jitter, and loss – audio is more sensitive to delay and jitter, whereas signaling is
more sensitive to loss
...
From a practical standpoint, in terms of user experience, these fine points may matter in some
cases and not in others
...
In this case the larger
signaling packets do not disrupt audio flow, because the serialization delay is low and because there are
so few signaling packets relative to audio packets
...

Suppose, however, that the ratio of signaling to audio is much greater – perhaps nearly 1:1
...

Suppose also that the WAN link is relatively small (typically less than 768k) and serialization delay is a
factor
...
It would be
advisable in this case to use separate queues, optimized for the different characteristics of audio and
signaling
...


KW

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The preceding paragraphs are generalizations, and are not meant to imply a firm set of rules
...
The explanations given here are
intended to give the reader a starting point
...


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Other Examples
Example 3
Suppose that C-LANs 192
...
1
...
11 cannot mark their traffic (pre-Communication Manager system)
...

access-list 101 permit ip host 192
...
1
...
168
...
0 0
...
0
...
168
...
11 192
...
2
...
0
...
255
Access list 101 permits any IP traffic from the two C-LANs to the 192
...
2
...

There is an implicit deny any at the end of this access list
...

Example 4
This is the same as example 2, but with more restrictions on the traffic
...
A somewhat matching set of configurations is applied to both routers
...
168
...
0 0
...
0
...
168
...
0 0
...
0
...
168
...
0 0
...
0
...
168
...
0 0
...
0
...

There is an implicit deny any at the end of this access list
...
16
...
1
apply the voipQoS policy outbound on this interface
service-policy output voipQoS
If any of the endpoints were incapable of DSCP marking, the “dscp 46” could be removed from access list 101
...


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Appendix G: IP Trunk Bypass – TDM Fallback Q&A
Q1:
How does the IP trunk bypass (aka TDM fallback) feature work, and how should the parameters be set on
the system-parameters ip-options form? How do these settings affect the IP trunk bypass feature?
The system-parameters ip-options form is used to define the thresholds that trigger a fallback to a TDM
trunk, thus bypassing the IP trunk
...
Simply
stated, a near-end MedPro/MR320 monitors network performance by pinging the far-end C-LAN to measure
network response against the configured thresholds
...
The VoIP module in the G700 does not behave exactly like the MedPro/MR320 board, and it
cannot perform the ping functions that a MedPro or MR320 performs
...

When a high threshold is reached the signaling group goes into bypass state, and a fallback TDM trunk is
utilized
...
Because networks and user preferences vary, there is no single set of optimal thresholds
...
The parameters are as follows
...
Many users begin to notice performance degradation at around 200-250ms oneway delay
...
100-150ms or less one-way delay typically results in very acceptable audio
quality
...
Avaya Labs testing
has shown that audio quality is acceptable even with 5% packet loss
...

Ping Test Interval (sec): This is the frequency at which pings are sent out
...
In loads prior to Avaya
Communication Manager 2
...
As of Communication Manager
2
...
1-2 sec is a good starting point for this parameter
...
10 should be used here for a minimum ping test interval of 10sec,
which results in calculations every 100sec to detect a network outage
...
1 and MedPro firmware v70, 20 to 30 pings at 1-second intervals results in
calculations every 20 to 30 seconds, which provides the granularity required to gauge
network performance
...
To facilitate this it is important to know that the call server can select any MedPro/MR320 in the
near-end system’s network region to originate the pings
...

Q2:
Besides the IP trunk bypass feature, what other mechanisms are in place to detect an outage or severe
congestion in the IP network, and how long does it take to detect it?

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See section 3
...
For the IP trunk as a whole the best method is the IP trunk bypass feature
...
This function assesses the IP trunk every 15 minutes in a G3r or
Linux platform, and every hour in a G3i platform
...
It can detect a network outage, but it does not assess
network performance
...
3
...
Assuming the failure
to set up the signaling link is the result of a network outage, the Maintenance Function detects this and puts the
signaling group out of service within one minute
...
There is an outage in the IP network between the two systems and the S8700
discovers this after a measurement interval (IP trunk bypass feature)
...
The S8300/media-gateway normally does not detect the outage until
the next Maintenance Function cycle
...
So the S8300 detects the outage less than
one minute after the first call attempt
...
In the case of severe congestion the S8700 detects the
congestion and puts the signaling group in bypass state, the same as with a network outage
...
(This message is also sent in the network outage case, but it doesn’t reach the
far end because of the outage
...
The same command at the S8300 shows the signaling group in far-end bypass state
...


Q3:
As a follow-up to the previous question, what are the effects of the two sides not detecting the outage at
exactly the same time?
Both sides accept incoming calls on TDM trunks, regardless of the state of IP trunks
...
Side B
continues to attempt using the IP trunk until it detects the outage, at which time it utilizes the TDM trunk for its
outbound calls
...
This causes side B to go to far-end bypass state and also use the TDM trunk
...


Q4:
When the IP network recovers after an outage or severe congestion, do both sides discover this at the same
time and start sending calls over the IP trunk at the same time? If not, what are the effects?
No, as with detecting the failure, detecting the recovery is also independent
...
So if side A detects the IP network
recovery first and calls side B while B is still in bypass state, side B accepts the call
...

The scenario for severe congestion is the same
...
If one of the points fails the IP trunk goes out of service
almost immediately at the local system where the failure occurred
...
At the remote system (the other end of the IP
trunk) the IP trunk eventually goes out of service as follows
...
The Maintenance Function, either at the normal
interval or triggered by a call attempt, puts the signaling group out of service
...
A way to compensate
for this type of outage is to administer multiple IP trunks (signaling groups and trunk groups) across multiple CLANs between the same systems
...
As long as there is at least one
MedPro/MR320 or VoIP module at each end with available DSP resources, the IP trunk is unaffected by
MedPro/MR320 or VoIP module failures
...
This essentially results in a bypass condition where the TDM trunk is utilized
...
They
discover if a C-LAN they are registered with has gone down, and re-home to a different C-LAN
...
323 and H
...


Q8:

How is call processing affected in general by a MedPro/MR320 or VoIP module outage?

The call server knows when a MedPro/MR320 or VoIP module has gone out of service and stops directing
calls to that device
...
If there is an outage during an active call, and that call is going through the
affected MedPro/MR320 or VoIP module, that call loses audio
...


Q9:

How is call processing affected in general by an IP trunk outage?

If the IP trunk outage is the result of a C-LAN/S8300 failure, direct IP-IP calls remain up until one of the IP
phone goes on hook
...
If the IP trunk outage is the result of the IP network going down, the audio is lost on active
calls, and new calls are routed over the fallback TDM trunk if one is administered
...
General VoIP requirements are discussed in the
AVAYA IP Voice Quality Network Requirements document available on the www
...
avaya
...
The following list summarizes key QoS requirements for voice traffic
...

Voice bearer and Voice Signaling Packet loss should not be greater than 3%
...

Average one-way jitter should be less than 30 milliseconds
...
323 Signaling Traffic should be marked DSCP 26 (AF31)
IPSI Call signaling traffic should be given a guaranteed bandwidth on WAN links
...


The IPSI circuit pack provides enterprises with the capability to IP-connect Port Networks over
LAN/WAN links in simplex and high availability configurations
...

The call signaling traffic is encapsulated AVAYA proprietary CCMS (Control Channel Message Set)
messages inside TCP/IP packets
...
323, H
...
931 messages used for
registration of IP endpoints, to setup and teardown calls, periodic testing of the hardware, and keep-alive
messages for IPSI connected port networks
...

The following table and graph displays IPSI call signaling traffic for varying Busy Hour Call Completion
rates (BHCC)
...
The simulated call scenario is a general business case
...

BHCC
Per PN
1K
2
...
5K
10K

KW

Usage Per Station
Light Traffic
Moderate Traffic
Heavy Traffic

IPSI Bandwidth (Kbps)
full duplex
17
...
5
52
...
8
83
...
5K

5K

7
...
The amount of overhead per VoIP call includes:
• Ethernet adds a 18 byte header, plus a 4 byte CRC plus an optional 4-byte 802
...

• Point-to-Point Protocol (PPP) adds 12 bytes of layer 2 overhead per packet
...

• Frame Relay adds 6 or 7 bytes per packet
...

• IPSI encryption adds up-to 23 bytes (AES) for the encryption header and padding in addition to
Layer 2 overhead
...
For example; for a busy hour call completion rate of 5K calls (moderate general
business traffic rate), the L2 overhead for a PPP link would be 61 PPS X 12 bytes/packet or 6
...
1 Kbps
...
2 Kbps for a total of 69
...

A general rule of thumb for IPSI Control traffic bandwidth allocation is to add an additional 64Kbps of
signaling bandwidth to the minimum required bandwidth in order to manage peak (burst) traffic loads
and either round up or down to nearest DS0
...
3Kbps + 64Kbps)
for IPSI signaling bandwidth across the WAN link
...
4 Kbps bandwidth on the standby link
...

BHCC
1K
1K w/ encryption
2
...
5K w/ encryption
5K
5K w/ encryption
>=7
...
5Kw/ encryption

Ethernet
64Kbps
64Kbps
128Kbps
128Kbps
128Kbps
128Kbps
192Kbps
192Kbps

PPP
64Kbps
64Kbps
128Kbps
128Kbps
128Kbps
128Kbps
192Kbps
192Kbps

MLPPP
64Kbps
64Kbps
128Kbps
128Kbps
128Kbps
128Kbps
192Kbps
192Kbps

Frame Relay
64Kbps
64Kbps
128Kbps
128Kbps
128Kbps
128Kbps
192Kbps
192Kbps

Cisco CBWFQ allows you to specify the exact amount of bandwidth to be allocated for a specific class of
traffic
...
Bandwidth can be assigned a percentage of total link speed or in Kbps
...
Optional WRED can selectively discard lower priority traffic
when the interface begins to get congested
...

Implement a QoS policies that provides:
-

A queue for IPSI traffic using, for example, DSCP 36 (AF42) for IPSI signaling traffic
...

Expedited Forwarding - DSCP 46 - like behavior for the real-time voice
...

Assured Forwarding (AF31) like behavior for H
...


Note: DSCP 34 (AF41) is reserved for Video in the Cisco AutoQoS model but can be assigned for IPSI
traffic when video is not deployed
...
The set diffserv 36 CLI command is used to mark traffic from IPSI to Server when you login
to the IPSI
...

It is important to note that H
...
Consult your account team for additional traffic requirements
...
, “Cisco IP Telephony Network Design Guide,” www
...
com, Customer Order
Number: DOC-7811103=, Copyright 2001
...
, “Cisco IP Telephony QoS Design Guide,” www
...
com, Customer Order
Number: DOC-7811549=, Copyright 2001
...
, “Configuring Compressed Real-Time Protocol,” www
...
com, July 2002
...
, “Troubleshooting Cisco Catalyst Switches to Network Interface Card (NIC)
Compatibility Issues,” www
...
com, July 2002
...
, “Understanding Compression (Including cRTP) and Quality of Service,”
www
...
com, July 2002
...
, “802
...
iee
...

[7] IETF, “RFC 2508: Compressing IP/UDP/RTP Headers for Low-Speed Serial Links,” www
...
org,
February 1999

---

Title: Avaya, Avaya Media Gateways, VOIP Protocols.
Description: Avaya Communication Manager, Avaya Media Gateways, implementation of VOIP set up. VOIP Protocols description & functions. i.e. H.323, H.248, CCMS.

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