Key features for entry level voip telephone SK655
The SK655 is a budget hotel SIP IP Phone for next generation network. It features high-quality speakerphone technology, and includes a call hold / transfer / 3-way conference, and 20 speed-dial buttons for various voice services.
The SK655 can effortlessly deliver excellent voice quality equivalent to the regular PSTN connections utilizing cutting-edge Quality of Service, echo cancellation, comfort noise generation (CNG), Automatic Gain Control (AGC) and voice compensation technology.
The SK655 is based on SIP V2.0 and compatibility with most service providers. It features single 10/100Mbps Ethernet ports with the IEEE802.3af PoE support ( Optional ).
• No LCD
• 20 speed dial keys
• Function Keys: HOLD / XFER / CONF / SPEEKER
• Single 10/100Mbps Ethernet port
• Compliant with IEEE 802.3af Power over Ethernet (PoE)
• Support AGC(Automatic Gain Control)/ VAD (Voice Activity Detection)/ CNG (Comfort Noise Generation)/ Echo
• Cancellation and Jitter buffer for excellent voice quality
• Very Highly affordable and effective IP Phone CPE
SK655 Specifications
| Hardware Features | |
| Interface | 1 RJ-45 WAN port(10/100Mbps Ethernet port) 1 DND switch Full duplex hands-free speaker phone Wall Mount 1 Power set |
| Power Supply | AC Input : 100-240v, 50-60Hz; DC Output: 5V, 1A IEEE802.3af PoE in WAN port(IP601 only) |
| Environmental | Operation Temperature: 5~45 Degree C Storage Temperature: -25~85 Degree C Relatively Humidity: 10%~90% No Condensing Shock: Up to 75cm Drop upon package |
| Dimension | (L)185×(W)140×(H)60mm |
| Unit Weight | 338g(without package) |
| Software Features | |
| Audio Codec Features | G.711(A-law, u-Law)with PAMS above 4.3 G.729A/AB with PAMS above 4.0 G.723: 5.3kbps and 6.3kbps G.722(high quality) Adaptive Jitter Buffer Management Voice Activity Detection Comfort Noise Generation Echo Cancellation |
| Protocols | SIP V2 (RFC 3261,3262,3263,3264) Backward Compatible with RFC2543 Session Timer (RFC4028) SDP (RFC2327) RTP/RTCP (RFC1889 and RFC1890) NAPTR for SIP URI Lookup (RFC2915) STUN (RFC 3489) ARP/RARP (RFC 826/903) SNTP (RFC 2030) DHCP/PPPoE HTTP Server for Web Management TFTP/HTTP for Auto Provisioning Message Waiting Indicator(RFC3842) DNS/DNS SRV (RFC1706 and RFC 2782) IEEE802.1Q VLAN/802.1p and DSCP |
| Management | Firmware Upgradable Web Management Interface Local and Remote Syslog (RFC3164) Auto Provisioning: TFTP, HTTP SNTP Time Synchronization Two User Level Configuration SNMPv2 TR069 |
| Applications | MAC Address Cloning SIP proxy redundancy: dynamic via DNS SRV, A records Direct IP to IP calling NAT Traversal: STUN QoS with Layer 2 and Layer 3 DHCP Client IP conflict detection Support PoE comply with IEEE802.3af(optional) |
| Call Features | 3-way Conference Music on hold DTMF Relay: In-band, Out-band and SIP INFO Call Hold Call Forwarding Call Mute Call Transfer Call Waiting Speed Dial(20 records) Full-duplex Speakerphone Delayed hotline Dial Plan MWI DND Volume Adjustment: Handset, Speaker and Ring Phone Features Customized Ring Tone Call History -Most Recently Missed Calls -Most Recently Received Calls -Most Recently Dialed Numbers Black List |