With versatile and robust architecture,Synway's SBC2000S Software Session Border Controller offers a complete connectivity solution for large enterprises and service provider and enables scalable, reliable and secured connectivity between diverse VoIP networks. Scaling up to 2,000 concurrent sessions, the SBC2000S connects IP-PBXs to any SIP trunking and cloud-based services, and offers superior performance in connecting any SIP to SIP environment.
The SBC2000S could be customized to multiple voice channels in cloud platform or in on-premise server to enable versatile connectivity between VoIP networks, such as connecting IP-PBX systems to any IP-based applications.SBC2000S rich features make this device an excellent option for mobile VAS, SIP trunking, contact center and emergency service deployments, as well as for retail, wholesale & business.
The SBC2000S could be customized to multiple voice channels in cloud platform or in on-premise server to enable versatile connectivity between VoIP networks, such as connecting IP-PBX systems to any IP-based applications.SBC2000S rich features make this device an excellent option for mobile VAS, SIP trunking, contact center and emergency service deployments, as well as for retail, wholesale & 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 |