Software Features:

zRFC1889 RTP: A Transport Protocol for Real-Time Applications

z ARP

Address Resolution Protocol

zRFC2327 Session Description Protocol

zRFC2543 SIP: Session Initiation Protocol

zRFC2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals

zRFC2880 Internet Fax T.30 Feature Mapping

zRFC2976 The SIP INFO Method

zRFC3261 SIP: Session Initiation Protocol

zRFC3262 Reliability of Provisional Responses in Session Initiation Protocol (SIP)

zRFC3263 Session Initiation Protocol (SIP): Locating SIP Servers

zRFC3264 An Offer/Answer Model with Session Description Protocol (SDP)

zRFC3265 Session Initiation Protocol (SIP)-Specific Event Notification

zRFC3311 The Session Initiation Protocol (SIP) UPDATE Method

zRFC3325 Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks

zRFC3362 Real-time Facsimile (T.38) – image/t38 MIME Sub-type Registration

zRFC3389 Real-time Transport Protocol (RTP) Payload for Comfort Noise (CN)

zEFC3489 STUN - Simple Traversal of User Datagram Protocol (UDP)

Through Network Address Translators (NATs)

zRFC3428 Session Initiation Protocol (SIP) Extension for Instant Messaging

zRFC3515 The Session Initiation Protocol (SIP) Refer Method

zRFC3550 RTP: A Transport Protocol for Real-Time Applications

zRFC3581 An Extension to the Session Initiation Protocol (SIP) for Symmetric Response Routing

zRFC3665 Session Initiation Protocol (SIP) Basic Call Flow Examples

zRFC3824 Using E.164 numbers with the Session Initiation Protocol (SIP)

zRFC3841 Caller Preferences for the Session Initiation Protocol (SIP)

zRFC3842 A Message Summary and Message Waiting Indication Event Package for the

Session Initiation Protocol (SIP)

zRFC3891 The Session Initiation Protocol (SIP) "Replaces" Header

zRFC3892 The Session Initiation Protocol (SIP) Referred-By Mechanism

zRFC3960 Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP)

zRFC4028 Session Timers in the Session Initiation Protocol (SIP)

Voice

zSIPv2 (RFC3261) compliance

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

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

zSIP Extension: Session Timer , Proxy-Require,

zMD5 Authentication for SIP

zSIP NAT Keep Alive Time

zFlash Hook Timer

zMWI- Message Waiting Indicator Tones

zVMWI- Via FSK

zStreaming Audio Server- UP to 10 Sessions

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

zCNG and VAD

zSilence suppression & detection

zG.165/G.168 Echo Cancellation

zAdaptive jitter buffer

zProgrammable gain control

zIn-band DTMF

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

zDTMF, Pulse dial support

zTermination Impedance: 600/900& complex Impedance

zFailover SIP Proxy server registrations

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

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

zTelephone book

zHot Line and Warm Line Calling

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Abocom DVG-2101SP manual Software Features, Voice

DVG-2101SP specifications

The Abocom DVG-2101SP is a versatile and powerful device designed primarily for digital voice applications and high-speed internet connectivity. This VoIP adapter stands out due to its ability to bridge the gap between traditional telephone systems and modern IP networks, making it an essential tool for businesses and home users transitioning to Voice over Internet Protocol technology.

One of the main features of the DVG-2101SP is its two FXS (Foreign Exchange Station) ports. These ports enable users to connect standard analog telephones or fax machines directly to the VoIP network, facilitating seamless communication without the need for additional devices. The integration of these ports allows users to maintain their existing phone infrastructure while benefiting from the cost savings associated with VoIP services.

In addition to its telephony capabilities, the DVG-2101SP offers a comprehensive firewall and security features. These include NAT (Network Address Translation) support, SIP (Session Initiation Protocol) proxy functionality, and various QoS (Quality of Service) settings that ensure optimal performance for voice calls. The device prioritizes voice packets over other types of data to guarantee clear audio quality, even during peak internet usage times.

Another significant aspect of the DVG-2101SP is its support for various voice codecs, including G.711, G.723.1, and G.726. This flexibility in codec support accommodates a wide range of bandwidth conditions and user preferences, making it suitable for various deployment scenarios. Whether in a low-bandwidth environment or a high-definition audio setting, the device can adapt to deliver the best possible experience.

User convenience is also a priority, as the DVG-2101SP features an intuitive web-based interface that simplifies configuration and management. This makes it easier for users to set up their device, manage settings, and troubleshoot issues without requiring extensive technical knowledge. Additionally, the DVG-2101SP supports remote management, allowing service providers to offer support and maintain devices without physical access.

Lastly, the compact design of the DVG-2101SP ensures it fits easily into any environment, whether in a small home office or a larger business setting. Its reliable performance, feature-rich design, and compatible technologies make the Abocom DVG-2101SP an excellent choice for anyone looking to upgrade their voice communication capabilities and enhance their internet experience.