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Abocom VOIP DVG-2101SP User Manual

Page 2

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Software Features:


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RFC1889 RTP: A Transport Protocol for Real-Time Applications

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ARP Address Resolution Protocol

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RFC2327 Session Description Protocol

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RFC2543 SIP: Session Initiation Protocol

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RFC2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals

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RFC2880 Internet Fax T.30 Feature Mapping

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RFC2976 The SIP INFO Method

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RFC3261 SIP: Session Initiation Protocol

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RFC3262 Reliability of Provisional Responses in Session Initiation Protocol (SIP)

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RFC3263 Session Initiation Protocol (SIP): Locating SIP Servers

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RFC3264 An Offer/Answer Model with Session Description Protocol (SDP)

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RFC3265 Session Initiation Protocol (SIP)-Specific Event Notification

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RFC3311 The Session Initiation Protocol (SIP) UPDATE Method

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RFC3325 Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted

Networks

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RFC3362 Real-time Facsimile (T.38) – image/t38 MIME Sub-type Registration

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RFC3389 Real-time Transport Protocol (RTP) Payload for Comfort Noise (CN)

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EFC3489 STUN - Simple Traversal of User Datagram Protocol (UDP)

Through

Network

Address

Translators

(NATs)

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RFC3428 Session Initiation Protocol (SIP) Extension for Instant Messaging

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RFC3515 The Session Initiation Protocol (SIP) Refer Method

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RFC3550 RTP: A Transport Protocol for Real-Time Applications

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RFC3581 An Extension to the Session Initiation Protocol (SIP) for Symmetric Response Routing

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RFC3665 Session Initiation Protocol (SIP) Basic Call Flow Examples

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RFC3824 Using E.164 numbers with the Session Initiation Protocol (SIP)

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RFC3841 Caller Preferences for the Session Initiation Protocol (SIP)

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RFC3842 A Message Summary and Message Waiting Indication Event Package for the

Session Initiation Protocol (SIP)

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RFC3891 The Session Initiation Protocol (SIP) "Replaces" Header

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RFC3892 The Session Initiation Protocol (SIP) Referred-By Mechanism

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RFC3960 Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP)

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RFC4028 Session Timers in the Session Initiation Protocol (SIP)

Voice


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SIPv2 (RFC3261) compliance

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SIP Proxy Redundancy: Dynamic via DNS SRV, A Records Re-registration with Primary SIP Proxy Server.

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Secure(Encrypted) Calling via Pre-Standard Implementation of Secure RTP.

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SIP Extension: Session Timer , Proxy-Require,

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MD5 Authentication for SIP

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SIP NAT Keep Alive Time

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Flash Hook Timer

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MWI- Message Waiting Indicator Tones

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VMWI- Via FSK

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Streaming Audio Server- UP to 10 Sessions

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Voice Compression: G.711 a/u-law , G.726, G.729A/B, G.723.1

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CNG and VAD

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Silence suppression & detection

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G.165/G.168 Echo Cancellation

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Adaptive jitter buffer

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Programmable gain control

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In-band DTMF

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Out-of-band DTMF relay: RFC2833/ SIP info

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DTMF, Pulse dial support

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Termination Impedance: 600/900& complex Impedance

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Failover SIP Proxy server registrations

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T.30 FAX pass through, T.38 real time FAX relay

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Caller ID: DTMF, FSK-Bellcore, FSK-ETSI detection and generation

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Telephone book

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Hot Line and Warm Line Calling