Phần mềm SBC Mediant VE/SE

Phần mềm Mediant VE (SBC) được thiết kế để đáp ứng nhu cầu ngày nay của các doanh nghiệp và nhà cung cấp dịch vụ muốn ảo hóa cơ sở hạ tầng của họ và hài hòa các trung tâm dữ liệu của họ trên phần cứng máy chủ.

Max. Signaling Sessions: 24,000
Max. Media Sessions: 24,000
Max. SRTP-RTP Sessions: 10,000
Max. Transcoding: 12,000 (MTC)
Max. Registered Users: 75,000

Mã: MediantVE/SE Danh mục: Thiết bị SBC Từ khóa: SBC Mediant

Thông số kỹ thuật

MODEL	Mediant VE
Capacities
Max. Signaling Sessions	24,000
Max. Media Sessions	24,000
Max. SRTP-RTP Sessions	10,000
Max. Transcoding Sessions	12,000 *
Max. Registered Users	75,000
Security
Access Control	DoS/DDoS line rate protection, bandwidth throttling, dynamic blacklisting
VoIP Firewall	RTP pinhole management, rogue RTP detection and prevention, SIP message policy, advanced RTP latching
Encryption and Authentication	TLS, DTLS, SRTP, HTTPS, SSH, client/server SIP Digest authentication, RADIUS Digest
Privacy	Automatic topology hiding, user privacy
Traffic Separation	VLAN/physical interface separation for multiple media, control and OAMP interfaces
Intrusion Detection System	Detection and prevention of VoIP attacks, theft of service and unauthorized access
Interoperability
SIP B2BUA	Full SIP transparency, mature and broadly deployed SIP stack, stateful proxy mode
SIP interworking	3xx redirect, REFER, PRACK, session timer, early media, call hold, delayed offer
Registration and Authentication	User registration restriction control, registration and authentication on behalf of users, SIP authentication server
Transport Mediation	SIP over UDP/TCP/TLS/WebSocket/SCTP, IPv4 / IPv6, RTP / SRTP (SDES/DTLS)
Header Manipulation	Ability to add/modify/delete SIP headers and message body using advanced regular expressions (regex)
URI and Number Manipulations	URI user and host name 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.726, G.729A/B, GSM-FR, AMR-NB, AMR-WB (G.722.2), SILK-NB/WB, Opus-NB/WB
Signal Conversion	DTMF/RFC 2833/SIP, T.38 fax, packet-time conversion
WebRTC Gateway	Interworking between WebRTC devices and SIP networks. Supports WebSocket, Opus, VP8 video coder, lite ICE, DTLS, RTP multiplexing, secure RTCP with feedback
NAT	Local and far-end NAT traversal for support of remote workers
Voice Quality and SLA
Call Admission Control	Based on bandwidth, session establishment rate, number of connections/registrations
Packet Marking	802.1p/Q VLAN tagging, DiffServ, TOS
Standalone Survivability	Maintains local calls in the event of WAN failure
Impairment Mitigation	Packet Loss Concealment, Dynamic Programmable Jitter Buffer, Silence Suppression/Comfort Noise Generation, RTP redundancy, broken connection detection
Voice Enhancement	Transrating, RTCP-XR, Acoustic echo cancellation, replacing voice profile due to impairment detection, Fixed & dynamic voice gain control
Direct Media	Hair-pinning of local calls to avoid unnecessary media delays and bandwidth consumption while avoiding media anchoring
Voice Quality Monitoring	RTCP-XR, AudioCodes One Voice Operations Center (OVOC)
High Availability	SBC high availability with two-box redundancy, active calls preserved
Quality of Experience	Access control and media quality enhancements based on QoE and bandwidth utilization
Test agent	Ability to remotely verify connectivity, voice quality and SIP message flow between SIP UAs
SIP Routing
Routing Methods	Request URL, IP address, FQDN, ENUM, advanced LDAP, third-party routing control through REST API
Advanced Routing Criteria	QoE, bandwidth, SIP message (SIP request, coder type, etc.), Layer-3 parameters
Redundancy	Detection of proxy failures and subsequent routing to alternative proxies
Routing Features	Least-cost routing, call forking, load balancing, E911 gateway support, emergency call detection and prioritization
SBC Media Types	Audio\Video\Fax\Text\Message Session Relay Protocol (MSRP)\Binary Floor Control Protocol (BFCP)
SIPREC	IETF standard SIP recording interface, supporting both audio and video SBC sessions
Management
OAM&P	Browser-based GUI, CLI, SNMP, INI Configuration file, REST API, HTTP reverse proxy
OAM&P	One Voice Operations Center (OVOC)
Multi-tenancy	Advanced multi-tenant SBC partitioning
Deployment tools	VNFM/Stack manager (Mediant CE), HEAT templates, Cloud Formation
Auto-scaling (CE)	Automatic, REST API, CLI, Web UI
Cloud Environments
Public cloud	Azure, AWS
Private cloud	OpenStack, VMware® vSphere