Thiết bị Ribbon SBC 1000 Gateway

Thiết bị Ribbon SBC 1000 Gateway cung cấp cho các doanh nghiệp vừa và nhỏ một hệ thống thoại an toàn, đáng tin cậy để kết nối VoIP với đầy đủ nhà cung cấp dịch vụ SIP. SBC 1000 hỗ trợ kết nối bất kỳ giữa các thiết bị hoặc thiết bị đầu cuối FXS, FXO, BRI, PRI và SIP, cho phép nhanh chóng di chuyển sang các mạng dựa trên SIP trong khi vẫn giữ các tổng đài hoặc thiết bị tương tự hiện có.

Số lượng cuộc gọi được chuyển đổi TDM/FXx <-> SIP tối đa: 128.
Số lượng cuộc gọi đồng thời tối đa: 192

Mã: SBC1000-Gateway Danh mục: Thiết bị Voice Gateway Từ khóa: Ribbon Gateway

MODEL	SBC 1000™ Gateway
System Capabilities
Sessions	Maximum number of TDM/FXx <-> SIP calls: 128
Sessions	Maximum number of total concurrent calls: 192
Call Set-Up	Maximum call set-up rate: 4 cps
Encryption	Maximum number of TLS sessions: 192
Encryption	Maximum number of SRTP sessions: 128
Business Continuity	Bypass relays (PRI <-> PRI, FXS <-> FXO) for emergency calls (e.g., 911)
	Site survivability for SIP clients (including Yealink® Teams and Polycom® UC phones and conference bridges) through a built-in SIP registrar
	PSTN fallback when WAN is down
	Multiple SIP trunking service provider support for redundancy
	ITSP E911 Support
	Rapid Ethernet Port Fail-over, to maintain inprogress calls in the event of an Ethernet port or switch problem
	Multiple Spanning Tree Protocol, to prevent network routing loops
	911 Call Preemption
	Microsoft® Office 365® Phone System E911 support; SIP/PIDF-LO pass through and ELIN Gateway
Media Services
	G.711, G.722, G.722.2 (AMR-WB), G.723.1 (5.3 kbps, 6.3 kbps), G.726 (32 kbps), G.729A/B (8 kbps), T.38, SILK-NB/WB
	T.38 with CNG tone detection
	DTMF/RFC4733; Inband DTMF; SIP INFO/RFC-2834
	Voice Activity Detection (VAD)
	G.168 Echo Cancellation with standard 128 ms tail length
	Comfort noise generation and packet loss concealment
	Automatic call type detection — voice, fax or modem
	Music on-hold
	Call progress tones — ringback, busy, re-order
	RTP inactivity monitoring (dead call detection)
	RTP pass-through and media bypass
	RTCP/RTCP-XR
	Caller ID support
Signaling
	Maximum number of signaling groups: 101
	TDM Signaling (ISDN): AT&T 4ESS/5ESS, Nortel DMS-100, Euro ISDN (ETSI 300-102), QSIG, NTT InsNet (Japan), ANSI National ISDN-2 (NI-2)
	TDM Signaling (CAS): T1 CAS (E&M, Loop start); E1 CAS (R2)
	Back-to-Back User Agent (B2BUA)
	SIP (UDP/TCP/TLS) <-> CAS/PRI/BRI/FXS/FXO
	CAS/PRI/BRI/FXS/FXO <-> CAS/PRI/BRI/FXS/FXO
	SIP (UDP/TCP/TLS) <-> SIP (UDP/TCP/TLS)
	FXS long loop support (up to 3km)
Protocol Support
	SNMPv2c, SNMPv4
	HTTPS
	SIP (RFC 3261) over UDP, TCP, TLS
	RTP/RTCP (RFC 3550, 3551)
	RTP/RTCP multiplexing over single UDP port (RFC 5761)
	DNS
	IPv4, IPv6, and IPv4/IPv6 interworking
	RIPv2, OSPF as dynamic IP routing protocols
	DHCP server
	DHCP client
	Asynchronous DNS for SIP
	NAT
	Support for Reason Header
Routing/Policy
	Interactive Connectivity Establishment (ICE), RFC 8445
	– Full implementation support, including connectivity check generation
	– Lite support, for public Internet ICE agents
	Maximum number of call route entries: 1,000
	Routing based on quality metrics
	Least cost routing
	Time of Day Routing
	Event-based action set
	On-board call forking (up to eight end points)
	Supplementary services
	– Call hold
	– Call transfer (blind & assisted)
	– Call forward
	Embedded policy/routing engine
	Optional centralized policy/routing via Ribbon Centralized
	Policy Server (PSX Server) using SIP
	Screening, blocking, routing, presentation, call type filters
	Route prioritization
	Leading digit routing; International routing; URI-based routing
	Digit manipulation (name/number manipulation)
	One number fax support (single DID for voice and fax)
	SIP routing
	– Based on source and destination IP address
	– Fully Qualified Domain Name (FQDN)
	Detect proxy failure and route to alternate paths
	Re-route on failure based on full Cause Code re-routing on T1/E1 trunks
Security
	TLS (Transaction Layer Security) for signaling encryption
	– TLS 1.2 (RFC 5246)
	– DTLS version 1.2 (RFC 6347)
	Secure Real-time Transport Protocol (SRTP) & Control
	Protocol (SRTCP) for media and media control encryption
	– SDES (Session Description Protocol Security Descriptions) key negotiation (RFC 4568)
	– DTLS extension for SRTP/SRTCP (RFC 5764)
	Built-in VoIP firewall
	Multiple unique X.509 public key certificates/PKCS #12 files (up to 11)
	Wildcard certificate support
	Topology hiding; User privacy
	Prevention of Denial-of-Service (DoS) and Distributed DoS (DDoS) attacks
	Dialed Number Identification Service (DNIS), Calling Line Identification (CLID), Call type pre-authentication
	Traffic separation (VLAN interface separation)
	Malformed packet protection
	Access Control Lists (ACLs)
	NAT/NAPT and port forwarding, NAT traversal
Quality of Service (QoS)
	Bandwidth management
	Call Admission Control (CAC) (deny excessive calls based on static configuration for bandwidth management)
	P-time mediation for rate limiting
	Per-call statistics
	Diffserv/DSCP marking
Microsoft® Teams®
	Certified SBC for Phone System Direct Routing
	Supports multiple tenant-related Direct Routing deployments with Microsoft partners/PSTN carrier
Microsoft Skype® for Business
	Qualified Enhanced Gateway for Lync® 2013, Lync 2010 & Skype for Business deployments
	Skype for Business, Lync 2013, and Lync 2010 qualified SBC and E-911 ELIN Gateway
	Microsoft Office 365® Exchange® Unified Messaging qualified
	Microsoft SCOM support
Additional Hardware Specifications
Front Panel	Additional dual USB 2.0 ports for ASM (optional)
	WAN and LAN Interfaces
	– 3 x 10/100/1000 BASE-T Ethernet ports with VLAN support
	– Auto-MDIX
	Physical PSTN Interfaces
	– Up to 2 T1/E1 CAS/PRI digital ports
Rear Panel	Physical PSTN Interfaces
	– Up to 2 T1/E1 CAS/PRI digital ports
	– Up to 4 BRI digital ports
	– Up to 24 FXS (Foreign eXchange Station) analog ports
	– Up to 4 globally compliant FXO (Foreign eXchange Office) analog ports
Available Feature Options	SIP <-> SIP signaling and media services, including transcoding, IPsec, and video passthrough / SIP registrations
	Voice quality reporting
	Call forking
	Azure® and on-premises Active Directory® Integration
	Routing protocols
	Skype for Business Quality of Experience (QoE) feature
	BroadSoft® BroadWorks® local survivability

Thương hiệu	Ribbon

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