Phần mềm SBC Mediant CE

Phần mềm Mediant CE (SBC) tận dụng các lợi thế của tính linh hoạt của cloud để cho phép các doanh nghiệp và nhà cung cấp dịch vụ nhận được một môi trường ảo có khả năng nhanh chóng điều chỉnh theo nhu cầu. Mediant CE tự động cung cấp thêm dung lượng khi có yêu cầu và thu nhỏ lại khi nhu cầu giảm. Kiến trúc microservice của nó, kết hợp với cụm phương tiện có thể mở rộng, cho phép các dịch vụ truyền thông tạo doanh thu mới được giới thiệu đơn giản và tiết kiệm chi phí.

  • Max. Signaling Sessions: 40,000
  • Max. Media Sessions: 40,000
  • Max. SRTP-RTP Sessions: 40,000
  • Max. Transcoding: 30,000
  • Max. Registered Users: 130,000
AvailabilityIn stock
Mediant CE
0,00 ₫
Thương hiệu:
MODEL Mediant CE
Max. Signaling Sessions 40,000
Max. Media Sessions 40,000
Max. SRTP-RTP Sessions 40,000
Max. Transcoding Sessions 27,000
Max. Registered Users 100,000
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
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
OAM&P Browser-based GUI, CLI, SNMP, INI Configuration file, REST API, HTTP reverse proxy
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
More Information
Thương Hiệu Audiocodes
SKU Mediant CE
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