Voice Features
- G.711 a-law 64K
- Packet Interval: 20/30/40 ms
- Concurrent Calls: 32 ch @ 20 ms
- G.711 μ-law 64K
- Packet Interval: 20/30/40 ms
- Concurrent Calls: 32 ch @ 20 ms
- G.723.1 5.3K/6.3K
- Packet Interval: 30/60/90 ms
- Concurrent Calls: 32 ch @ 30 ms
- G.726 32K
- Packet Interval: 20/30/40 ms
- Concurrent Calls: 32 ch @ 20 ms
- G.729 8K
- Packet Interval: 20/30/40 ms
- Concurrent Calls: 32 ch @ 20 ms
- DTMF Detection and Generation
- Silence Suppression & Detection
- Comfort Noise Generation (CNG)
- Voice Activity Detection (VAD)
- Echo Cancellation (G.165/G.168)
- Adaptive (Dynamic) Jitter Buffer
- Call Progress Tone Generation
- Auto or Programmable Gain Control
- Built-in Local Mixer
- ITU-T V.152 Voice-band Data over IP Networks
SIP Call Features
- Peer to Peer Call
- Call Hold / Retrieve
- Call Waiting
- Call Pick Up
- Call Park / Retrieve (SIP Server Required)
- Call Forward - unconditional, busy, no answer
- Call Transfer - attended, unattended
- Do Not Disturb
- Speed Dialing
- Repeat Dialing
- Three-way Calling
- MWI (RFC-3842)
- Hot Line and Warm Line
Telephony Specifications
- In-Band DTMF, Out-of-Band DTMF Relay (RFC2833 or SIP INFO)
- DTMF / PULSE Dial Support
- Caller ID Generation / Detection:
- DTMF
- FSK-Bellcore Type 1 & 2
- FSK-ETSI Type 1 & 2
- FSK-NTT
- FSK: Calling Name, Number, Date and Time, VMWI
- FXS Metering Pulse:
- Polarity Reversal
- 12 kHz calling tone
- 16 kHz calling tone
- T.30 FAX Bypass to G.711, T.38 Real Time FAX Relay
- FXS Line test and diagnostics with visual alarm
- indication
- Inward self test:
- Loopback - codec
- Loopback - analog
- SLIC DC power voltage
- Tip / Ring DC feed
- Ringer
- Outward Test (GR909 Standard) :
- REN
- Phone Line disconnected
- H.F. DC Voltage (Hazardous and foreign DC Voltage)
- H.F. AC Voltage (Hazardous and foreign AC Voltage)
- Tip / Ring Short
- Modem over IP up to V.34
- ROH Tone (Receiver Off-Hook Tone @ 480 Hz)
- Loop Current Suppression
SIP Account Management
- By Port Registration
- By Device Registration (share account)
- Mixed Mode (Hunt number for inbound, by port number for outbound)
- Invite with Challenge
- Register by SIP Server IP Address or Domain Name
- Support RFC3986 SIP URI Format
SIP Call Management
- Support Outbound Proxy
- Register up to three SIP servers
- SIP Registration Failover Mechanism
- Group Hunting
- Privacy Mechanism / Private Extensions to SIP
- Session Timers (Update / Re-invite)
- DNS SRV Support
- Call Types: Voice / Modem / FAX
- Call Routing by Prefix Number
- User Programmable Dial Plan Support
- CDR Client
- Manual Peer Table (for P2P calls)
- E.164 Numbering, ENUM support
IP Network Specifications
- Support IPv4, IPv6 future upgradable (Option)
- WAN: Static IP, PPPoE, DHCP, PPTP
Network Protocol Support:
IP, TCP, UDP, TFTP, FTP, RTP, RTCP, ARP, RARP, ICMP,
NTP, SNTP, SNMP v1/v2, HTTP, HTTPS, DNS,
DNS SRV, Telnet, DHCP Server, DHCP Client,
STUN Client, UPnP, IGMP snooping, IGMP proxy
QoS Support:
WAN: DiffServ, IP Precedence, Priority Queue,
Rate Control, 802.1Q (VLAN Tagging), 802.1p (Priority
Tag)
LAN: Rate Limit
DDNS Support
Network Security Specifications
- VPN PPTP Client
- DIGEST Authentication
- MD5 Encryption
- DoS Protection
Management
- Web-based Configuration
- Auto-provisioning (HTTP / HTTPS)
- Telnet
- IVR
- FTP / TFTP / HTTP Software Upgrade
- Configuration Backup and Restore
- Reset to Default Button
- TR-069/104 (Option)
SIP, Voice and FAX Related Standard
- RFC1889 RTP: A Transport Protocol for Real-Time Applications.
- RFC2543 SIP: Session Initiation Protocol
- RFC2833 RTP Payload for DTMF Digits, Telephony
- Tones and Telephony Signals
- RFC2880 Internet Fax T.30 Feature Mapping
- RFC2976 The SIP INFO Method
- RFC3261 SIP: Session Initiation Protocol
- RFC3262 Reliability of Provisional Responses in
- Session Initiation Protocol (SIP)
- RFC3263 Session Initiation Protocol (SIP): Locating SIP Servers
- RFC3264 An Offer/Answer Model with Session Description Protocol (SDP)
- RFC3265 Session Initiation Protocol (SIP) - Specific Event Notification
- RFC3311 The Session Initiation Protocol (SIP) UPDATE Method
- RFC3323 A Privacy Mechanism for the Session Initiation Protocol (SIP)
- RFC3325 Private Extensions to the Session Initiation
- Protocol (SIP) for Asserted Identity within Trusted Networks
- RFC3362 Real-time Facsimile (T.38) - Image/t38 MIME Sub-type Registration
- RFC3515 The Session Initiation Protocol (SIP) Refer Method
- RFC3550 RTP: A Transport Protocol for Real-Time
- Applications. July 2003
- RFC3665 Session Initiation Protocol (SIP) Basic Call
Flow Examples
- RFC3824 Using E.164 numbers with the Session
- Initiation Protocol (SIP)
- RFC3842 A Message Summary and Message Waiting
- Indication Event Package for the Session Initiation
- Protocol (SIP)
- RFC3891 The Session Initiation Protocol (SIP)
- “Replaces” Header
RFC3892 The Session Initiation Protocol (SIP) Referred-By Mechanism
RFC3960 Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP)
RFC3986 Uniform Resource Identifier (URI): Generic Syntax
RFC4028 Session Timers in the Session Initiation Protocol (SIP)
Draft-ietf-sipping-service-examples-08 for call features
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