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Softswitch ECSS-10

Key benefits

Functionality of private branch exchange, rural, city, trunk line, combined and international telephone exchanges
Virtual PBX
Call-center function
Large conference service
Geographical redundancy
Active-active redundancy mode
Scalability
User-friendly management interface

ECSS-10 is a hardware and software platform that is designed for integrated infocomm networks construction. The software and hardware components of ECSS-10 were developed and manufactured by Eltex and have a high level of reliability.

Application
4/5 class Softswitch ECSS-10 is a flexible system for any level communication center construction: departmental net-works, enterprise networks and all the classes of provider networks (local, zonal, transit, intercity, international).

Key features

100 000 + subscribers
AutoProvision
Certified as private branch exchange, rural, city, trunk line, combined and international telephone exchanges
Virtual PBX
Call-center
Large conference service
Operation as SaaS platform
Support for Session border controller functions
Support for a wide range of VAS (Value Added Services)
Group notification
Support for Astra Linux
Geographic redundancy
Local redundancy
Hot software update
Load balancing
Flexible IVR builder
Support for widely used CRM, integration with client’s CRM
Operation under KVM and VM Ware

Scalability
Scalability is driven by ECSS-10 modular architecture. It allows using the solution in small corporate communication centers as well as in international transit stations.

Virtual PBX
ECSS-10 supports Virtual PBX service. It allows connecting subscribers to dedicated Virtual PBX with private dial plans, modern services, billing reports, etc. The clients obtain up-to-date VoIP services without additional expenses for installing and maintaining of a standalone PBX.

Mobile VoIP
Mobile IP telephony provides telecommunication services to remote subscribers and helps to optimize roaming costs.
Via the Internet, mobile VoIP service can be used anywhere in the world due to the use of secure connection through session border controllers. For accessing this service, you need to install SIP application on your mobile device and activate an account.
Through the FMC service, a subscriber mobile phone can be connected to a corporate network without the Internet connection. In this case, all the data will be transmitted via GSM channels.

Monitoring and management
A unified monitoring and control interface implemented via Eltex.EMS system provides customer with easy-to-use management tools: network elements aggregation, centra-lized configuration and firmware version management, scheduled maintenance, main parameters monitoring in a single window.

AutoProvision
Autoprovision subsystem is used for centralized configuring of phones and VoIP gateways. The subsystem allows uploading configuration files to end-user devices automati-cally. The wide range of vendors is supported: Eltex, Yealink, Cisco, Grandstream, Snom, Siemens, Fanvil, etc.
AutoProvision subsystem provides automatic configuration for different user accounts and transparent replacing of a phone (not only to other model, but to other vendor device as well). After replacing a phone, the configuration will be adapted automatically.

Fault tolerance
The cluster architecture of ECSS-10 Softswitch allows achieving 99.9999% of reliability. Due to the local active-active redundancy and geographic redundancy, any single hardware failure won’t be able to affect calls at any stage of their processing.

SIGTRAN
The SIGTRAN subsystem implemented on ECSS-10 supports MTP3, ISUP and M2UA. The redundancy system for signal and media traffic is fully supported. H.248/MEGAGO are used as gateway management protocols. Supported transport protocols are SCTP, UDP and TCP.

Supported protocols*

SIP 2.0 (RFC 3261)
H.248/Megaco
RADIUS AAA
ISUP
MTP3
SIP-T/SIP-I
T.38
SNMP
M2UA

Protocols supported via gateways*

Signaling system №7
R1.5, R2
EDSS-1/Q.931
V5.1, V.5.2
CAS

Supported audio codecs*

G.729A/B
G.711A/U
G.726
G.723.1 (5.3, 6.3 kbps)
G.722, G.722.1, G.722.1c
GSM FR
iLBC
Speex
L16
AMR
OPUS

Supported video codecs*

H.263-1998
H.264

Management

The main means that provide efficient management and access rights delimitation:

MML console (SSH)
Web 2.0 interface http(s)
Subscriber portal — customizable web interface for subscriber’s VAS management
Customizable web interface for Virtual PBX management
Web interface for large conference management

Redundancy

Support for hardware redundancy
Local redundancy in active-active mode
Hot-swappable software modules
Support for geographic redundancy

Telephone routing*

Routing by mask
The route selection based on the parameters:
- Calling party number (CgPN)
- Calling party category (CPC)
- Called party number (CdPN)
- Subscriber group ID
- Nature of address (NOA)
- Numbering Plan (NP)
- Calling Party Address Presentation Restricted Indicator (Calling Party APRI)
- Week day
- Time of day
- Gateway/direction load levels
- By a tag
- Redirecting number
- Original Called Party Number
- On the presence of a number in number list
Numbers modification
Flexible management of calls processing by a graphical scenario
Call-center organization, flexible routing among queues
Support for external routing via RADIUS and HTTP

Large conference service

200+ participants in conference
Support for conference templates
Support for VoIP SIP phones and extension panels
Configuring of private and public conference templates
Easy-to-use improved web interface
Conference planning
The number of available active large conferences defines by a license

Call charging

RADIUS Accounting
CDR files

Call-center

Corporate phonebook
An agent can work with a phone only (without a PC)
Web workstation with wide function set for calls control/processing
Supervisor web workstation for call-center monitoring
Call-center settings management in the web interface
A wide range of call distribution algorithms
Smart prediction of queuing time
Call statistics collecting and reporting
Call prioritization
Call distribution according to agent’s qualification
Collection of subscribers’ feedback on call center agents performance
Queue hierarchy
Call pickup from a queue
Supervisor driven manual mode for calls distribution in a queue
Support for Callback feature in a queue

Additional functions

Support for different media resources formats
SIP Registrar
Authentication via LDAP and/or RADIUS
Session border controller functionality
Secure media streams using SRTP
Connecting subscribers with incompatible codecs
IPv6 for modern IP networks
Text to Speech/Speech to Text engine for services
Renewed subscriber web interface
Email2Fax and Fax2Email services
Reconfigurability — the opportunity to increase performance and functionality
Integration with Microsoft Active Directory
Text message transmission
Trunk based black and white lists
Trunk based CPS limiting
Load balancing among several media servers
Location based media traffic routing
Phonebook synchronization
Direct RTP forwarding mode
Voice traffic recording
Support for Distinctive ring and Distinctive picture
Caller ID characters encoding conversion**
Multi-user video conference (up to 5 subscribers) through internal conference server
Transscrambling — speech to text conversion engine
Integration with CRM: AmoCRM/Megaplan
Requesting external catalogues of subscribers/companies for substitution of Displaying names/companies according to phone number
Multicast IP Paging, Multicast IP Listen
Forwarding of external and local calls to different phone numbers

Value Added Services (VAS)

Call group services
Call Hunt
Group call with optional hidden redirection (CGG)
Boss group
Zone Page
Group Pickup
Auto Attendant
Direct Inward Dialing
Subscriber services
Calling Line Identification Presentation (CLIP)
Calling Line Identification Restriction (CLIR)
Calling Line Identification Restriction Override (CLIRO)
Calling Name Identification Presentation (CNIP)
Calling Picture Identification Presentation (CPIP)
Customizing Ring Back tone
Personal voice notification/intercom (Voice Page)
Call intervention
Distinctive Picture
Distinctive Ring
Anonymous call barring (ACB)
Call forwarding: unconditional, busy, no response, out of service, time based, based on Caller ID (CFU, CFB, CFNR, CFOS, CFT, CFCID)
Hidden call forwarding (HideCFName)
Call forwarding barring: outgoing calls (FBC), incoming calls (RFC)
Find me (forwarding to a group of number) unconditional, time dependant, Find me on No Response
Follow me, Follow me No Response
Call forwarding through subscriber SIP terminal (CFSIP)
Call Waiting service (CWAIT)
Multiline
Call Hold (CHOLD)
Call Pickup
Call transfer (CTR)
Call park
3-way Conference
Conference call, Add-on
VIP calls
Walkie-talkie mode
Selective call acceptance, Incoming Whitelist
Selective call rejection, Incoming Blacklist
Selective call Origination, Outgoing Whitelist
Selective call Origination, Outgoing Blacklist
Missed call notification
VAS management via a phone
Call recording
Video Call Recording
Voice Mail
Message Waiting Indication (MWI)
SIP Presence and Busy Lamp Field
Call back
Auto Redial, Auto Redial with Call back
Redial
Speed Dial
Do Not Disturb (DND)
Alarm call
Malicious call Identification (MCID)
My number
Outgoing calls barring (RBP)
Hotline/Warmline
Authorization on a remote phone (Remote phone)
Duplicating incoming calls to additional internal or external number (FlexiCall)
Introducing message for called subscriber (Introduce)
Second handset
Smart cancel
Subscriber IVR script
Privacy — intervention barring
Adjustable remote ring tone
Fax-to-Email, Email-to-Fax

Firmware version 3.14

* The lists might be extended upon a request