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EZSIP-6000 SIP Communication Server
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An affordable high available redundant SIP proxy server which support dual IPv4/IPV6 stacks
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| EZSIP-6000 SIP Communication Server is a powerful and scalable SIP call processing system (SIP Proxy Server) which provides built-in rich telephony service at affordable price. It is a software module running under Linux 64 bits operation system and support IPV4 and IPV6 dual stacks simultaneously. The hitless redundant feature increases reliability and high availability of SIP Communication Server EZSIP-6000.
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Highlights and Benefit
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• Support both IPv4 and IPv6 SIP Calls Simultaneously
• Support Hitless HA Redundant
• SIP UDP, TCP, TLS Seamless Access
• Automatic Audio/Video NAT Detection and Traversal
• Support SIP Trunk and SIP Router
• Flexible yet Powerful Digit Processing and Call Routing Plan
• Easy Web Management and System Morning
• Prosperous Telephony Features for Time to Market
• Detect Potential SIP Attacks and Prevention
• IP Country/Network Lock
• Optional Voice Logging Module
• Support RFC-8599 Push Softphone
• Support SRTP Transcode
• CPE Auto Provisioning
• Running under Off-the-Shelf Server and 64 bits Linux
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Product Specifications
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| System Requirements |
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• INTEL/AMD CPU (Intel® 64)
• RHEL 8/Rocky Linux 8
• MYSQL Database
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| Protocols |
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• SIP RFC 3261
• SIP UDP, TCP, TLS
• RTP, SRTP, RTCP
• IPV4 / IPV6 Dual Stack
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| System Service |
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• Hitless HA Redundant
• Automatic TCP and UDP Traversal
• Automatic IPV4 and IPV6 Traversal
• Support Multiple SIP Domain
• Automatic Audio/Video NAT Traversal
• SIP Proxy/Registrar
• Support Permanent & Dynamic Contact
• Support SIP Trunking
• Support LAN/WAN Access Simultaneously
• RADIUS Billing Support
• Extension/Device Monitoring
• Device Allowance Control
• Session Timer Call Validation
• INVITE-Initiated Dialog Event Package (RFC 4235)
• External Voice Mail Server Support
• Missed Call Email Notice
• CPE Auto Provisioning
• In Call Service
• Support RFC-8599 Push Softphone
• Support SRTP Transcode
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| Routing Plan |
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• Group Based Routing
• Time of Day Routing
• Preference Routing
• Round Robin Routing
• Load Balancing Routing
• Broadcast Routing
• Unavailable Redirect
• ENUM Routing
• Black List Rejecting Route
• ANI Based Route
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| SIP Attack Detection and Protect |
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• SIP Attack Detection/IP Blocking
• SIP User Device Restriction
• Country/IP Network Lock
• Enhanced Password Option
• Black Routing List
• CAPTCHA to Protect Web
• Web Access Log
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| System Capabilities |
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• Max Concurrent Users: 30,000
• Max Busy Hour Call Completion: 270,000
• Max Concurrent Calls: 5,000
• Max NAT Resource: 2,000
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| Telephony Features |
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• DID/DOD
• Call Transfer
• Call Hold
• Call Waiting
• Call Forward
• Call Display Name
• Call Pickup (Group, Specified, Global)
• Calling Line Identification Presentation (CLIP)
• Calling Line Identification Restriction (CLIR)
• Digit Manipulation
• Local Emergency Call Group
• Secondary PSTN Number
• Parallel Hunting for Multiple Contacts
• Follow Me Always
• Time of Day Follow Me
• Incoming Call Blocking
• Outgoing Call Blocking
• Outgoing Privilege Calling
• Do Not Disturb
• Anonymous Call Blocking
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| Voice Logging (optional) |
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• Max Logging Channels: 512
• Support Selectable Logging Target or Number
• Support G.711, GSM, G.729, G.722 and iBLC decode
• MP3 Compress with Optional AES encryption
• Support Recording On Demand/div>
• Separate Caller/Called Voice Channel
• Provides Voice Logging Detail Record
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| Management |
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• Multi-Language Support
• Web Provisioning Access Log
• Easy Web Management (HTTP/HTTPS)
• 3 Levels of Access Controlling Rights
• Customizable Web Access Right
• SIP Attack Detection and Prevention
• System Alert through SYSLOG and Email
• SOAP Provisioning Interface
• Real-time Status & Tracing
• Scheduled Update Task
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Screenshots
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| • Home Page |
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| • SIP Service |
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| • System Security |
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| • Routing Plan |
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| • SIP Trunk |
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| • Voice Logging Report |
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| • Creating an Extension |
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| • Call Features |
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