Support SIP 2.0, TCP/UDP/IP, RTP/RTCP, HTTP, HTTPS, ICMP, ARP, DHCP, NTP/SNTP, TFTP protocols. Support STUN and symmetric RTP for NAT. Interoperable with various 3rd party SIP end user device, Proxy/Registrar Server, and gateway products. Advanced 32bit embed MCU + DSP technology to ensure superior audio quality. Advanced adaptive jitter buffer control, packet delay and loss concealment technology. Support popular vocoders including G.723, G.729A, G.711(a-law and u-law), G.726(40K/32K/24K/16K). Support standard voice feature such as Caller ID Display, Call Waiting, Hold, Transfer, in-band and out-of-band DTMF, three way conference. Support Silence Suppression, VAD(Voice Activity Detection), CNG(Comfort Noise Generation), Line Echo Cancellation(G.168), and AGC(Automatic Gain Control) PSTN pass through port, with auto switch between PSTN and VOIP on inbound call and manual switch on outbound call. It becomes PSTN phone when unit power is off. ATA-1100 analog phone adapter is a small CPE. It is easy to install. 1. One phone(FXS) RJ-11 jack interface. 2. One line(FXO) RJ-11 jack interface 3. One Ethernet 10baseT RJ-45 jack interface. 4. One unit status LED 5. One data link and activity LED. 6. One PSTN connection status LED 7. One VOIP connection status LED. 8. One PSTN and VOIP switch button. For more details, please welcome to view our websit