The Proxy7™
Signaling Gateway
uniquely combines a SIP
proxy server with SS7/C7 signaling intelligence
to create the most effective bridge between
emerging VoIP networks and the PSTN. An
integral part of sentitO's ONX architecture,
the Proxy7 works in conjunction with sentitO's
IVG™ and third-party application servers
to help global service providers reduce
the costs and complexities of integrating
VoIP and legacy signaling networks.
The Proxy7 simplifies and
reduces the cost of integrating VoIP networks
with the PSTN by eliminating signaling backhaul
requirements for carriers using SS7 IMTs.
In addition, the Proxy7 manages a single
point code and distributes that code across
multiple IVGs to simplify the management
of wholesale and retail customers.
The Proxy7 scales to support
over 500,000 BHCA in a single chassis and
natively supports over 40 international
signaling variants to ensure carrier-grade
deployment of VoIP services around the world.
With enhanced routing capabilities, the
Proxy7 allows carriers to quickly and cost-effectively
expand their VoIP service offerings.
For carriers using SS7 IMTs,
the Proxy7 translates ISUP messages into
SIP and performs number translation centrally,
instead of on individual voice gateways.
This greatly reduces the complexity of deploying
distributed VoIP networks that scale to
support growing customer bases.
| PROXY
SPECIFICATIONS |
| SYSTEM REQUIREMENTS |
Sun Fire V120
Sun Netra V120
Supports Dual Processor V120 for
higher performance
|
| CAPACITY |
500,000 BHCA
per system
Supports up to 24 SS7/C7 links
Supports one point code
|
| REDUNDANCY |
1:1
In service Software Upgradeability
|
| REGISTRATION
AND AUTHENTICATION |
Digest
MDS
|
| CONFIGURATION |
CLI via RS-232
Interface and Telnet SNMPv1 via
Previsio™ GUI |
| ACCOUNTING AND
MANAGEMENT |
SNMP Traps
Detailed Logging to Text Files
Call Detail Records
|
| SS7 / C7 FEATURES |
Stateful Routing
ANSI, ITU and Country Variants
Cause Code Mapping
Glare Control
Number Translation
Continuity Checking
Policy-Based Call Routing
Toll-Free / 800 Calls
Calling Name (CNAME) Query
Local Number Portability (LNP)
|
| SS7
/ C7 STANDARDS COMPLIANCE AND
CERTIFICATION |
| MTP |
ANSI T1.111.1-T1.111.2
(1996)
ANSI T1.111.3 (1996)
ANSI T1.111.4-T1.111.8 (1996)
ITU Q.701 (1993)
ITU Q.702 (1989)
ITU Q.703-Q.704 (1997)
ITU Q.705-Q.706 (1993)
ITU Q.707-Q.708 (1989)
|
| SCCP |
ANSI T1.112.x
(1992)
ITU Q.711-Q.714 (1993)
|
| TCAP |
ANSI T1.114 (1992)
ANSI T1.114 (1996)
ITU Q.771-Q.775 (1993)
ITU Q.771-Q.775 (1997)
|
| ISUP |
ANSI T1.113.x
(1992)
ANSI T1.113.x (1996)
ITU Q.761-Q.764 (1993)
ITU Q. 761-Q.764 (1997)
|
| TELCORDIA |
GR-246
GR-317
GR-394
GR-905
GR-1298
GR-1299
GR-1428
GR-1432
GR-2936
GR-2982
TR-1188
TR-0533
|
| SIP
STANDARDS COMPLIANCE |
RFC
3261: Session Initiation Protocol
RFC 3262: Reliability of Provisional
Responses in SIP
RFC 3263: SIP: Locating SIP Servers
RFC 3264: An Offer Answer Model
with Session Description
RFC 3265: SIP—Specific Event
Notification
RFC 3311: The SIP Update Method
RFC 3515: The REFER Method
RFC 2806: URLs for Telephone Calls
RFC 2915: The NAPTR DNS Resource
Record
RFC 2976: The SIP INFO Method
RFC 3581: An Extension to the
Session Initiation Protocol for
Symmetric Response Routing
draft-ietf-sipping-cc-transfer-05:
Call Transfer
draft-ietf-sip-referredby-00:
The Referred-By Mechanism
draft-ietf-sip-replaces-03: The
SIP Replaces Header
draft-ietf-sip-session-timer-10:
The SIP Session Timer
draft-ietf-sipping-mwi-02.txt:
A Message Summary and Message
Waiting Indication Event Package
for the Session Initiation Protocol
draft-yu-tel-url-07: Extension
to Tel URL to Support Number Portability
and Freephone Service
|
|
|
