Sessions Border Controller (SBC)
SBC120H (E1/SIP)
120 Sessions SBC Featuring High Security, SIP Normalization, Transcoding, Effortless Diagnosis
120 Sessions SBC Featuring High Security, SIP Normalization, Transcoding, Effortless Diagnosis
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.
This device supports voice densities up to 120 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.
This device supports voice densities up to 120 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.
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 |