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Sessions Border Controller (SBC)

SBC1000H

SBC1000H (E1/SIP)

1000 Sessions SBC Featuring High Security, SIP Normalization, Transcoding, Effortless Diagnosis

  • 1000 Sessions & Registered Users
  • Signaling, RTP, Transcoding
  • Superior DSP Processing
  • Field-Proven Technologies
Rich Features and High Flexibility in Bosth PSTN and IP Networks

Configurable for E1/T1 VoIP gateway funtion, this SBC model supports voice densities in PSTN networks(E1/T1), call routing, call translation and IP transcoding for gateway operations. The integrated gateway functionality not only provides interworking between IP and TDM domains, but also automated failover from IP to TDM for outbound routing. These capabilities make this device an excellent option for mobile VAS, SIP trunking, contact center, service providers, enterprises & carriers.

Hybrid IP & PSTN Architecture Combined With Unparalleled Functionalities

This device supports voice densities up to 1000 channels (E1/T1), call routing, call translation and IP transcoding for gateway operations. The integrated gateway functionality not only provides interworking between IP and TDM domains, but also automated failover from IP to TDM for outbound routing. These capabilities make this device an excellent option for mobile VAS, SIP trunking, contact center and emergency service deployments, as well as for retail, wholesale and business.

Hybrid IP & PSTN Architecture Combined With Unparalleled Functionalities

This device supports voice densities up to 1000 channels (E1/T1), call routing, call translation and IP transcoding for gateway operations. The integrated gateway functionality not only provides interworking between IP and TDM domains, but also automated failover from IP to TDM for outbound routing. These capabilities make this device an excellent option for mobile VAS, SIP trunking, contact center and emergency service deployments, as well as for retail, wholesale and business.

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Features
High Interoperability
Adopted by over 500 SPs and enterprises, and proven interoperability with SIP trunks, SIP platforms and IP cloud services

Enhanced Security
Security-oriented, robust perimeter defense against cyber, DoS and DDoS attacks, as well as eavesdropping, fraud and service theft


High Resiliency
Telco-grade reliability, with High Availability (HA) using 1+1 active/standby redundancy, local branch survivability and PSTN fallback

Integrated Transcoding
Eliminate need to add separate hardware to support transcoding requirements, and support a range of Codecs, including G.723, G.729, G.711, iLBC, SIPINFO, RFC2833, RF3261, INBOUND and more.

Improved Integration
SBC bridges operators with enterprise communication networks seamlessly, rationalizes traffic, streamlines business procedures, and optimizes SIP trunking, PBX/H.323/SIP and VoIP applications.

Capabilities
Flexible Scalability
5 ~ 1000 simultaneous SIP sessions with dedicated DSP-powered multimedia transcoding provide high performance in a small footprint to help lower ownership cost and reassure high security at borders.

Deliver Hi-Flexibility
Secure connectivity and controlling can be offered by SBC1000H (E1/SIP), as it efficiently supports bilateral and multilateral interconnection among different networks; besides, Dialing and IP/PSTN routing available.

Superior Voice Quality
Integrate decades of SW/HW technologies to obtain advanced capabilities for optimizing and monitoring voice service quality

Loading Optimization
Load protection at the border of network provides highly differentiated voice and multimedia services through multi-level resource-balancing policies, registration limit, ID authentication and authorization.

Enhanced Security
SBC1000H (E1/SIP) offers real-time communication surveillance solution for enterprises/SPs/carriers communication systems, and helps them deliver any business-driven applications effortlessly and safely.
Core Specifications
Typical Features Dos/DDos protection; QOS/ TOS/DSCP setting; Signal encryption(TLS/IPSec); Media encryption(SRTP);
NAT transverse; SIP interworking; Support IPV4、IPV6 and VPN; Load balancing; Transmission speed limit; RTP encoding/decoding; Anti-phreaking; Redundancy and Backup
Security Access Control:    DoS/DDoS line rate protection, bandwidth throttling, dynamic blacklisting (Intrusion Detection System)
Encryption/Authentication:    TLS, SRTP, HTTPS, SSH, client/server SIP Digest authentication
Privacy:      Topology hiding, user privacy
Traffic Separation:    Self-adjustable automatic load balance
Intrusion Detection System:  Detection and prevention of VoIP attacks, theft of service and unauthorized access
VoIP firewall:  Optional
Interoperability
SupportedSIP B2BUA:    Full SIP transparency, mature and broadly deployed SIP stack, stateful proxy mode
SIP Interworking:     3xx redirect, REFER, PRACK, early media, call hold
Registration and Authentication:     User registration restriction control, registration and authentication on behalf of users, SIP authentication server for SBC users
Transport Mediation:       Mediation between SIP over UDP/TCP/TLS, IPv4/IPv6, RTP/SRTP
Header Manipulation:      Add/modify/delete SIP headers and message body using simple WireShark-like language with powerful capabilities such as variables and utility functions
Number Manipulations:       Ingress and egress digit manipulation
Transcoding and Vocoders:      Coder normalization including transcoding, coder enforcement and re-prioritization, extensive vocoder support: G.711, G.723.1, G.729, GSM-FR, AMR-NB, SILK-NB/WB, Opus-NB/WB
Signal Conversion:     DTMF/RFC 2833/SIP, T.38 fax, T.38 V3, V.34, packet-time conversion
NAT:               Hosted NAT, RTP self-adaption
WebRTC controller:     Optional or customizable
Voice Quality and SLA Call Admission Control:     Limit number and rate of concurrent sessions and registers per peer for inbound and outbound directions
Packet Marking:       802.1p/Q VLAN tagging, DiffServ
Standalone Survivability:    Maintains local calls in the event of WAN failure. Outbound calls can use PSTN fallback (including E911).
Impairment Mitigation:     Dynamic Programmable Jitter Buffer, Silence Suppression/Comfort Noise Generation
Voice Monitoring and Enhancement:   acoustic echo cancellation, fixed and dynamic voice gain control, dynamic programmable jitter buffer, silence suppression, RTP redundancy, broken connection detection
Direct Media:       Hair-pinning (no media anchoring) of local calls to avoid unnecessary media delays and bandwidth consumption
High Availability:     SBC high availability with 1+1 redundancy, active calls preserved
Test Agent:           Ability to remotely verify SIP message flow between SIP UAs
Echo cancellation:     G.168 128 ms tail length
Advanced Media Processing:       T.38 real-time fax, T.38 – G.711 interworking
SIP Routing Routing Criteria:    Incoming SIP trunk, DID ranges, host names, any SIP headers, codecs, QoE, bandwidth
Route To:   Configured SIP peers, registered users, IP address, request URI
Advanced Routing Features:   Alternative routes, load balancing, least-cost routing, call forking, E911 emergency call detection and prioritization
SIPREC:   SynAPI recording interface


Management

G.711A/U law, G.723.1, G.729A/B,G.726,iLBC,AMROAM&P:    Browser-based GUI, SNMP, INI Configuration file



Resources
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