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G860 Media Gateway Hardware
G860 Media Gateway Hardware
G860 Media Gateway Hardware
G860
Overview
Gateway supporting high density interfaces
9 to12 DS3 interfaces supported in the chassis
limited by CM capacities of 45-50K BHCC in first release.
OC3, STM-1 interfaces in future release
Small Footprint (5U) and Lower Cost
Significant reduction in number of PNs needed
SIP-based control
Carrier Grade Reliability (redundant system elements)
Element Management System from AudioCodes.
Ideal for Large IP-based Contact Centers and Campus
Target Market
Value
Significant reduction in equipment (gateway) price
Reduction of carrier access charges (break-even point for DS3 is
around 7 T1s)
Reduction in per-minute costs trough a carrier commoditization
strategy – FCE model
Lower costs associated with acquiring and maintaining data center
floor space (up to 85-90% reduction in space required)
Operational and maintenance cost savings by eliminating a large
number of G650 gateways (less power, UPS, cooling needs)
Installed and supported by Avaya Global Services
SIP gateway can provide SIP connectivity to other SIP-based
components (e.g., Avaya Voice Portal in consideration)
Example
Customer requires 3000 trunks (125 T1s)
Configuration with only G650 will need at the very least
16 Port Networks
32 G650s with High/Critical Reliability
8 Fully Occupied Racks (4 G650s/Rack)
Configuration with HDTG/IP Phones will need (approximately)
2-4 Port Networks or
4-6 G650s with High/Critical Reliability
3 Racks
Note 1: More G650s might be needed to support voice mail, digital/analog phones,
IVRs, announcements, queuing, etc. Footprint reduction will vary for each
customer.
Note 2: HDTG costs is comparable to two critically reliable G650 Port Networks
(gateways + IPSIs + Media Resource)
Positioning
Trunk Frac T1 (2) T1 (8) T1 DS3 (3) DS3
Capacity
G860
G650
Port
Network
G700
H.248
G350
H.248
G250
H.248
IXC/
IXC/
PSTN
PSTN
DS3
SIP
HDTG
Media (RTP)
Shuffling, Misc.
Shuffling is supported
From G860/SIP Trunks to any SIP Phone
From G860/SIP Trunks to any H.323 Phones registered to the
same CM
From G860SIP Trunks to any SIP Trunk
Shuffling is not supported
From G860/SIP Trunks to an H.323 Trunk
PSTN Trunk
ISDN PRI NFAS signaling
A-law, u-law
Design/Capacity Rules
A single Communication Manager can support estimated
45-50k Busy Hours SIP Calls
A single CLAN can support estimated 5k Busy Hour Calls
A single Communication Manager can support up to
5,000 SIP trunks (please refer to SIP trunk rules)
Multiple Port Networks will be required for
Time-slots and DSPs for queues, announcements, music-
on-hold, conference, etc
More IPSI throughput (to handle messages from all
CLANs)
Limitations
PRI-equivalent UUI transport not supported across SIP
BSR, CTI routed calls based on Adjunct Route with UUI,
UUI-related Call Center vectoring operations affected.
Resolution planned in the 4.1 release
DC Power only
AC version coming at the latest in 4.1
T.38 Fax, G.729A/AB and SRTP not supported in this
release
G729A/AB and SRTP support is there, just not tested
Testing planned for 4.1 release
No Modem support
Limitations
SIP
NMS
Browse EMS Client Client Nodes Map View
in MG Context &
Unified Alarm
View
SIP
SIP
AppServ/Pro
xy
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