ATL Telecom IP300S - Manual (Page 8)

IP SIP Phone v2 User’s Guide Mar. 2005
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z Support REGISTER, INVITE, ACK, CANCEL, BYE, OPTION, REFER, MESSAGE,
SUBSCRIBE, NOTIFY, INFO methods
z Support “alert-info” header for distinctive ring.
1.2.2. Real-time Voice Streaming
z Fully complies with RFC 1889 (RTP / RTCP), RFC 1890 (AVT profiles), RFC 3551 (RTP
Profile for Audio and Video Conference with Minimal Control) and RFC 3555 (MIME Type
Registration of RTP Payload Formats).
z Support both in-band DTMF mixed with RTP voice stream and out-of-band DTMF over RTP
(RFC2833).
z Dynamic RTP de-jitter buffer and lost packets concealment management.
z Speech CODEC supports: G.711 (A-law and µ-law), G.723.1/G.723.1A (both 5.3 and 6.4
kbps), and G.729A/G.729AB. CODEC precedence is configurable to adjust to your network
link speed.
z 3-way local conferencing
z Voice activity detection (VAD) and comfort noise generation (CNG).
z Voice and ringer volume control
z Real-time acoustic echo canceller.
z IP Type of Service (ToS) bits set for RTP/RTCP packet prioritization
z 802.1P precedence bits support to prioritize RTP voice frames within switched network.
z Configurable RTP / RTCP ports.
1.2.3. NAT and Firewall
z Support static NAT mapping (both NAT IP and SIP/RTP ports are configurable)
z Support Simple Traversal of UDP through NAT, (RFC 3489 STUN).
z Support auto-detect (auto-update) the change of NAT IP by STUN (in case the NAT has no
static IP and employ dial-up to public internet).
z draft-ietf-mmusic-sdp4nat-03.txt (RTCP attribute in SDP)
z Symmetric RTP flow for the cases where only one endpoint behind NAT.
z Support STUN server redundancy by DNS SRV/AAAA records
1.2.4. Management
z TFTP and HTTP for Auto-Provision
z HTTP configuration by web browser
z Key-pad configuration
z TELNET configuration