71 James Way
Eatontown, NJ 07724 USA
1-877-SPEAK-IP
1-732-460-9000
1-732-544-9119 (fax)
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Tenor and Quintum are registered trademarks. Call Relay SP, PacketSaver, Quintum Technologies, Inc.,
VoIP Made Easy, TASQ, SelectNet, and SelectNet Technology are trademarks of Quintum Technologies,
Inc.
•Chapter 3: Installation. Describes how to install the Call Relay 60 unit, including how to set
the IP address.
•Chapter 4: Getting Started with Configuration. Includes information for accessing the Command Line Interface (CLI) and Tenor Configuration Manager.
•Chapter 5: System Alarms. Describes how to use the Alarm Manager and tells you how to view
alarms via telnet session.
About this Guide
•Chapter 6: Diagnostic/Maintenance. Describes how to troubleshoot and monitor the health of
the system.
•Warranty/Approvals
•Glossary
•Index
P/N 480-0076-00-00 Preface-2
Typographical Conventions
Product Guide Conventions
Certain typographical conventions are used throughout this product guide. See below.
•All commands you enter via keystrokes appear in bold (e.g., Press Enter or Press Ctrl-I).
•All text commands you enter via Telnet session or command line typing appear in italics (e.g.,
type active).
•All references appear in italics (i.e., Choose File> Open).
•There are three types of special text that are designed to reveal supplemental information: Note,
Warning, and Caution. See below.
NOTE provides additional, helpful information. This information may tell you how to do a certain task or just be a reminder for how-to’s given in previous sections. (i.e., For a list of valid
commands at any time, type ?)
A WARNING provides information about how to avoid harm to your VoIP equipment or other
equipment. (i.e., Do not stack more than 4 units together.)
About this Guide
•A CAUTION provides information about how to avoid injury to yourself or to others. (i.e., Do
not install the equipment during a lightning storm.)
P/N 480-0076-00-00Preface-3
Finding Help
For additional help regarding configuration, see Chapter 4: Getting Started with Configura-
tion.
About this Guide
P/N 480-0076-00-00 Preface-4
Chapter 1: Overview
This chapter gives you a general overview of the Call Relay 60, including feature descriptions and
capabilities.
Specifically, the following topics are covered:
Tenor Call Relay description
Features
Capabilities
P/N 480-0076-00-001-1
Chapter 1: Overview
What is Tenor®Call Relay 60?
The Call Relay 60 Session Border Controller is a stand-alone unit which acts as an inter-domain
VoIP portal used to directly connect (over IP) one or more VoIP networks, supporting up to 60
simultaneous calls between networks. Call Relay 60 provides a single point for call control manage
ment, administrative services, call accounting, and security. All calls are switched through multiple
IP networks with just one single compression and decompression of the voice.
The unit connects with the IP network via 10/100 Mbps Ethernet connection; calls are transmitted
through the Ethernet LAN and routed over the corporate WAN or internet. It routes the call accord
ing to the parameters or defaults you have configured in the routing database.
Call Relay 60 features NATAccess, an intelligent network address translation technology. This
enables a VoIP network with multiple H.323/SIP endpoints to operate behind firewalls equipped
with Network Access Translation (NAT). NATAccess solves the problems commonly experienced
when using H.323/SIP devices behind firewalls; it eliminates the need to place these endpoints in
the “DMZ” or public IP domain, or to open the firewall to the point where serious security concerns
are raised. Call Relay 60 enables enterprises to expand their VoIP networks to home offices, branch
offices, customers, partners, and across the public Internet.
-
-
Figure 1-1 Call Relay 60
P/N 480-0076-00-001-2
Chapter 1: Overview
Features and Capabilities
State of the art Management system
The Call Relay 60 is managed by the Tenor Configuration Manager and Tenor Monitor. Through the
Tenor Configuration Manager, you can configure all options. An easy-to-use Java-based installation process enables you to install the manager and start configuring within minutes. Through the Ten or
Monitor, you can monitor the activity level and status of the system’s active calls, including alarms,
call detail records, etc. Both the Tenor Configuration Manager and Tenor Monitor provide compre
hensive on-line help systems that are available at your fingertips.
In addition, you can configure the unit through the Command Line Interface (CLI). Through this
simple telnet session, you can access all configuration options, including an online help system,
built into the CLI, which provides help for all features and functions. Just type help with the com
mand name at any prompt, and data about that field will be displayed.
Direct Connection Between VoIP Networks
As an inter-domain portal, the Call Relay 60 switch provides direct IP connection between multiple
networks. The switch eliminates the need to link two or more VoIP networks via PSTN gateways.
Figure 1-2.
See
-
-
A direct connection between VoIP networks provides lower latency and the best possible voice quality by eliminating multiple compression/decompression steps when linking multiple VoIP networks.
VoIP Network A
VoIP Network A
Call Relay60
Call Relay 60
VoIP Network B
VoIP Network B
Call Relay60
Call Relay 60
VoIP Network C
VoIP Network C
Figure 1-2
Direct IP Connection via Call Relay 60
IP Network
IP NetworkIP Network
VoIP Network D
VoIP Network D
Call Relay60
Call Relay 60
VoIP Network E
VoIP Network E
Call Relay60
Call Relay 60
VoIP Network F
VoIP Network F
Call Relay60
Call Relay 60
Call Relay60
Call Relay 60
P/N 480-0076-00-001-3
Chapter 1: Overview
NATAccess™/Remote NAT support
NATAccess™ is an intelligent network address translation technology. It enables H.323-based and/
or SIP based VoIP networks to operate behind firewalls equipped with Network Access Translation
(NAT); this provides maximum network security. NATAccess simplifies deployment and installation
by eliminating the need to place the Call Relay 60 on a public IP network. Using NATAccess pro
-
vides an easy, secure expansion between multiple VoIP sites.
Figure 1-3
“DMZ” LAN
Deploying the Quintum Tenor with NATAccess on the “Private” LAN
NATAccess
RouterRouter
“DMZ” Port
IP NetworkIP Network
“Public” LAN
Firewall/NATFirewall/NAT
“Private” Port
“Private” LAN
The Remote NAT feature enables Tenors with SIP applications to be used in an environment where
endpoints are behind NAT firewalls. As long as a Call Relay 60 is placed in the public IP domain, it
can communicate transparently with any VoIP endpoint (even from other vendors) that is located
behind a NAT firewall. In the case of a NAT firewall application, the actual IP address and port
number that the call comes from is the public (WAN) IP address of the NAT/firewall/router.
Figure 1-4
Remote NAT
Call Relay 60
Call Relay60
Call Relay 60
SIP Phone
Tenor
NAT/Router
NAT/Router
IP NetworkIP Network
NAT/Router
NAT/Router
PC SIP Client
T/A
Phone
P/N 480-0076-00-001-4
Chapter 1: Overview
Remove Unnecessary Call Delay
Since redundant decompression and re-compressions processes are eliminated as a result of Call
Relay 60 (linking two VoIP networks via PSTN gateways is not required); this improves voice qual
ity by removing unnecessary delay, latency, and distortion.
-
Figure 1-5
VoIP
VoIP
VoIP
Endpoint
Endpoint
Endpoint
VoIP
VoIP
VoIP
Endpoint
Endpoint
Endpoint
VoIP
VoIP
VoIP
Endpoint
Endpoint
Endpoint
Call Relay 60
Call RelayCall Relay
Call RelayCall Relay
VoIP
VoIP
VoIP
Endpoint
Endpoint
Endpoint
VoIP
VoIP
VoIP
Endpoint
Endpoint
Endpoint
VoIP
VoIP
VoIP
Endpoint
Endpoint
Endpoint
VoIP Network AVoIP Network B
VoIP Network AVoIP Network B
Removes Unnecessary Call Delay
Endpoint
Endpoint
Endpoint
IP Used to Link
IP Used to Link
Networks
IP Network
IP NetworkIP Network
Eliminates
Eliminates
redundant encoding
redundant encoding
and decoding
and decoding
Networks
Endpoint
Endpoint
Endpoint
Endpoint
Endpoint
Endpoint
Endpoint
Endpoint
Endpoint
Endpoint
Endpoint
Endpoint
Endpoint
Endpoint
Endpoint
VoIP
VoIP
VoIP
VoIP
VoIP
VoIP
VoIP
VoIP
VoIP
Call Relay 60
VoIP
VoIP
VoIP
VoIP
VoIP
VoIP
VoIP
VoIP
VoIP
Voice Control Management
H.323 Gatekeeper
The Call Relay 60 complies with the H.323 industry specifications for voice control and management. It performs IP call routing functions (for calls entering and exiting a site). The Gatekeeper
(internal to a Call Relay 60) collects, manages, and distributes call routing information. The Border
Element (internal to the Call Relay 60) provides access into or out of an administrative domain.
Other H.323 endpoints, such as gateways, can register to the internal Gatekeeper.
Session Initiation Protocol (SIP)
SIP (Session Initiation Protocol) is a signaling protocol used to establish a session on an IP network
for voice control and management; it is a request-response protocol that closely resembles Hypertext
Transfer Protocol (HTTP), which forms the basis of the World Wide Web. SIP re-uses many of the
constructs and concepts of Internet protocols such as HTTP and Simple Mail Transfer Protocol
(SMTP). The purpose of SIP is only to establish/change/terminate sessions. SIP is not concerned
with the content or details of the session.
SIP is Transport layer-independent, which means it can be used with any transport protocol: UDP,
TCP, ATM, etc. It is text-based, so it requires no encoding/decoding like H.323. And SIP supports
user mobility, using proxies and redirecting requests to your current location.
When configured for SIP the Call Relay 60 will act as a SIP User Agent (Endpoint) as defined in
IETF RFC3261. Multiple user agents allow for separate agents to be allocated to each SIP call. It
will be able to direct calls to and from the IP network, and Customer Premise Equipment (CPE) such
P/N 480-0076-00-001-5
Chapter 1: Overview
as phones, PBX's, and FAX machines, or the Public Switched Telephone Network (PSTN). The Call
Relay 60 SIP User Agent will work in conjunction with an external SIP proxy or redirect server to
route and connect calls over SIP based networks.
There are three basic components of SIP:
1. User Agent (Endpoint)
•client element, initiates calls
•server element, answers calls
2. Network Server (Proxy Server or Redirect Server)
•name resolution
•user location
•redirect and forking
3. Registrar
•Stores registration information in a location service using a non-SIP protocol.
H.323/SIP Signaling Translation
Through the Call Relay 60, there are two signaling options provided: H.323/SIP Signaling Translation and Signaling Gateway with Direct Media Connection. See below for a description of each.
H.323/SIP Signaling Translation. This enables the signaling and media (voice) to be sent from
the network (whether H.323 or SIP based) to the Call Relay 60 and out to another network (whether
H323 or SIP-based). The signaling is translated (from H.323 to SIP or vice versa) if necessary; this
simplifies inter-networking between diverse VoIP networks.
Figure 1-6
H.323 Based Network
H323/SIP Signaling Translation
3
2
3
a
.
i
d
H
e
M
IP NetworkIP Network
M
Call Relay 60
S
I
e
P
d
i
a
SIP Based Network
Signaling Gateway with Direct Media Connection. Enables signaling to be passed through
the Call Relay 60, then media (voice) is streamed directly between the network endpoints. In cases
where the origination network, the Call Relay, and destination network are in three different geo
-
graphical areas, this feature decreases delay time in voice media by sending it between the origina-
P/N 480-0076-00-001-6
Chapter 1: Overview
tion and termination networks directly, rather than having it run out of the origination network,
through the Call Relay 60 and back out to the destination network.
In this mode, the Call Relay 60 can support up to a maximum of 90 simultaneous calls when used as
an H.323 signaling gateway (this mode of operation is known as the “Gatekeeper Routed Call
Mode”); endpoints can register to the Call Relay 60’s gatekeeper.
Figure 1-7
Signaling Gateway with Direct Media Connection (for H.323)
Call Relay 60
H
H
.
3
.
2
3
3
2
/
3
S
I
P
VoIP Network B
VoIP Network A
P
I
3
S
2
/
3
3
.
2
3
H
.
H
IP NetworkIP Network
Media
Provides Load Balancing
Using multiple Call Relay 60 units provides a way for load balancing. For example, one unit can be
used as a primary portal and another as a secondary portal. You can divide the calls between portals
or simply use one as a backup. In the event of an IP link failure, the redundancy offers maximum
reliability.
SNMP Support
The Call Relay 60 supports Simple Network Management Protocol (SNMP), the standard protocol
used to exchange network information between different types of networks. The Call Relay 60 acts
as an SNMP agent to receive commands and issue responses to the network manager. The network
manager will then be able to perform certain functions, such as generating and sending traps.
Call Relay 60 communication
The Call Relay 60 provides a single IP address for accessing the public IP network.
Easy Connect to Console
Plugging a DB-9 cable provided by Quintum between a PC and the unit’s asynchronous RS-232 port
will connect the unit and get you up and running. Through this port, you are able to configure an IP
address for the Call Relay 60 portal.
P/N 480-0076-00-001-7
Chapter 1: Overview
Capabilities
Single Point of Network Interconnection
The Call Relay 60 aggregates all the traffic from the various IP endpoints and gateways within the
network and passes it to an endpoint at the “edge” of the another network, such as another Call
Relay 60 unit, IP phone or gateway. See
Figure 1-8.
Figure 1-8
VoIP Gateways
SD
Cisco AS5300
SERIES
VoIP Gateways
SD
Cisco AS5300
SERIES
Cisco AS5300
SERIES
Cisco AS5300
SERIES
Single Point for Network Access and Administration
Tenor VoIP Switch
SD
IPIP
SD
Call Relay SP
60
Call Relay SP
Tenor VoIP Switch
60
In each case, the Call Relay 60 provides a single IP address for entry to the complete network and
provides isolation between the networks in such a way that the internal structure of each remains
anonymous to the other. All voice calls and inter-network management function, including call man
agement, administration, and call accounting, to pass over the single inter-network IP connection. A
complete Call Detail Record (CDR) is generated by every call passing through the Call Relay 60 in
each direction. This simplifies the process of cross billing between customers. Each partner can
compile data on traffic entering and exiting their network; they are then able to generate and audit
inter-company billing.
A innovative method of linking VoIP networks is to use the Call Relay 60 approach to interconnect
the networks and perform call management. This also takes advantage of the Packet Saver feature,
which minimizes bandwidth usage.
-
P/N 480-0076-00-001-8
Chapter 1: Overview
PacketSaver™
PacketSaver packet multiplexing technology reduces the amount of IP bandwidth required to support multiple calls flowing between two networks. PacketSaver minimizes bandwidth usage by
aggregating samples from multiple VoIP conversations and packing them into a larger IP packet
with a single IP header. The process removes the need to send a bulky IP header with individual
voice samples. As a result, it eliminates the transmission of redundant information.
Figure 1-9 PacketSaver
Call A
Call B
Call C
Call A
Call B
Call C
Call Relay 60
Call Relay 60
Call A
Call B
Call C
Call A
Call B
Call C
Call Detail Recording
Through the Call Detail Recording (CDR) feature, the Call Relay 60 is able to generate a CDR at the
completion of each call. A CDR is a string of ASCII data which contains call information such as
call date and time, call length, calling party and called party. From this information you can capture
billing type data, which can be used to create billing reports.
P/N 480-0076-00-001-9
Chapter 1: Overview
Call Relay 60 with Tenor MultiPath Switch
See Figure 1-10 for an example of using Call Relay 60 with the Tenor VoIP MultiPath Switch. The
Tenor VoIP MultiPath switch will route calls coming from the Call Relay 60 unit, to either the PSTN
or PBX.
PABX
Analog Phones
PSTN
Figure 1-10
PSTN
PSTN
Tenor VoIP Switch
BRANCH OFFI CE
or TELEWORKER
IP
Call Relay 60
Call Relay 60
Swit ch
Router/ NAT/
Fir ewall
Call Relay
Router/NAT/Firewall
Router/NAT/Firewall
IP Phones
Swit ch
PC
Call Relay
IP Phones
IP Network
IP Network
60
P/N 480-0076-00-001-10
Chapter 2: Hardware Components
This chapter tells you what is contained in your hardware package. A description of each component
is also included.
Specifically, the following topics are covered:
Hardware
Cables
Specifications
P/N 480-0076-00-00 2-1
Chapter 2: Hardware Components
Hardware Description
Call Relay 60 is a stackable/rack mountable device which provides connections to two different
sites: Ethernet LAN and a PC.
Front Panel Connections and Reset/Diag Options
Figure 2-1 Front
Reset
Diag
View
LAN1/LAN2
Console Port
Reset. Resets the entire unit.
Diag. Enables you to perform software diagnostic procedures.
LAN 1/LAN 2 10/100 Base-T Ethernet ports. LAN 1 port provides an RJ-45 jack for an individual
connection to a 10/100 Ethernet LAN switch or hub via RJ-45 cable; it is individually configured
with a unique IP and MAC address. LAN 2 Ethernet port is reserved for future use.
Figure 2-2 10/100 BASE-T Ethernet Port Pin Order
Table 2-1 Input/Output 10/100 Ethernet port
Pin #SignalDefinition
1TX +Transmit Data
2TX - Transmit Data
3RX +Receive Data
4RSVDReserved
P/N 480-0076-00-002-2
Chapter 2: Hardware Components
Pin #SignalDefinition
5RSVDReserved
6RX -Receive Data
7RSVDReserved
8RSVDReserved
•Console port. This RS-232 connector is used for connection to a PC’s serial port via DB-9
serial cable at 38400 BPS 8 N 1, None. The input/output signals are listed in
Table 2-2.
Figure 2-3
DB-9 Female Connector Pin Order
5 4 3 2 1
9 8 7 6
Table 2-2 Serial RS232 DB-9 Connector Pinouts
Pin #FunctionDescription
1DTRData Terminal Ready
2TXDTransmit Data
3RXDReceive Data
4CDCarrier Detect
5GNDSignal Ground
6N.C.No Connect
7N.C.No Connect
8N.C. No Connect
9N.C.No Connect
P/N 480-0076-00-002-3
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