Grandstream Networks HT-496 user manual Sales@voipon.co.uk Tel +44 01245 808195 Fax +44 01245

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MTU A Maximum Transmission Unit (MTU) is the largest size packet or frame, specified in octets (eight- bit bytes), that can be sent in a packet- or frame-based network such as the Internet. The maximum for Ethernet is 1500 byte.

NAT Network Address Translation

NTP Network Time Protocol, a protocol to exchange and synchronize time over networks The port used is UDP 123 Grandstream products using NTP to get time from Internet

OBP/SBC Outbound Proxy or another name Session Border Controller. A device used in VoIP networks. OBP/SBCs are put into the signaling and media path between calling and called Caller. The OBP/SBC acts as if it was the called VoIP phone and places a second call to the called Caller. The effect of this behavior is that not only the signaling traffic, but also the media traffic (voice, video etc) crosses the OBP/SBC. Without an OBP/SBC, the media traffic travels directly between the VoIP phones. Private OBP/SBCs are used along with firewalls to enable VoIP calls to and from a protected enterprise network. Public VoIP service providers use OBP/SBCs to allow the use of VoIP protocols from private networks with internet connections using NAT.

PPPoE Point-to-Point Protocol over Ethernet is a network protocol for encapsulating PPP frames in Ethernet frames. It is used mainly with cable modem and DSL services.

PSTN Public Switched Telephone Network. The phone service we use for every ordinary phone call, or called POT (Plain Old Telephone), or circuit switched network.

RTCP Real-time Transport Control Protocol, defined in RFC 3550, a sister protocol of the Real-time Transport Protocol (RTP), It partners RTP in the delivery and packaging of multimedia data, but does not transport any data itself. It is used periodically to transmit control packets to participants in a streaming multimedia session. The primary function of RTCP is to provide feedback on the quality of service being provided by RTP.

RTP Real-time Transport Protocol defines a standardized packet format for delivering audio and video over the Internet. It was developed by the Audio-Video Transport Working Group of the IETF and first published in 1996 as RFC 1889

SDP Session Description Protocol is a format for describing streaming media initialization parameters. It has been published by the IETF as RFC 2327.

SIP Session Initiation Protocol, An IP telephony signaling protocol developed by the IETF (RFC3261). SIP is a text-based protocol suitable for integrated voice-data applications. SIP is designed for voice transmission, uses fewer resources, and is considerably less complex than H.323. All Grandstream products are SIP based

STUN Simple Traversal of UDP over NATs is a network protocol allowing clients behind NAT (or multiple NATs) to find out its public address, the type of NAT it is behind and the internet side port associated by the NAT with a particular local port. This information is used to set up UDP communication between two hosts that are both behind NAT routers. The protocol is defined in RFC 3489. STUN will usually work well with non-symmetric NAT routers.

TCP Transmission Control Protocol is one of the core protocols of the Internet protocol suite. Using TCP, applications on networked hosts can create connections to one another, over which they can exchange data or packets. The protocol guarantees reliable and in-order delivery of sender to receiver data.

TFTP Trivial File Transfer Protocol, is a very simple file transfer protocol, with the functionality of a very basic form of FTP; It uses UDP (port 69) as its transport protocol.

Grandstream Networks, Inc.

HT–496 User Manual

Page 32 of 33

 

Firmware 1.0.3.64

Last Updated: 1/2007

VoIPon www.voipon.co.uk

sales@voipon.co.uk Tel: +44 (0)1245 808195 Fax: +44 (0)1245 600030

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Contents Grandstream Networks, Inc Table of Contents Table of Figures Welcome Installation Definitions of the HT-496 ConnectorsFive Easy Steps to Install the HT-496 Able 2 HT-496 Technical Specifications Product OverviewHT-496 Hardware Specifications Basic Operations HT-496 IVR Menu DefinitionsReset Examples Bellcore Style 3-way Conference Expected outcomesStar Code Style 3-way Conference Call Features HT-496 Call Feature DefinitionsFlash/Hook Way conferenceAble 6 HT-496 LED Definitions Configuration Guide Static IP ModeUser Level Password Web pages allowed HT-496 Basic Configuration Settings Definitions Device Mode AccessDaylight Savings TimeHT-496 Device Status Page Definitions Advanced User Configuration HT-496 Advanced Configuration Page Definitions Caller ID Scheme On-hook VoltagePolarity Reversal NTP serverHT-496 Individual Account Settings Definitions Sales@voipon.co.uk Tel +44 01245 808195 Fax +44 01245 Fax Mode Lock keypad updateSpecial Feature Volume AmplificationSaving the Configuration Changes Software Configuration IVR MethodDirections Restore Factory Default Setting Directions for RestorationGlossary of Terms Sales@voipon.co.uk Tel +44 01245 808195 Fax +44 01245 Sales@voipon.co.uk Tel +44 01245 808195 Fax +44 01245 Sales@voipon.co.uk Tel +44 01245 808195 Fax +44 01245

HT-496 specifications

Grandstream Networks HT-496 is a cutting-edge Analog Telephone Adapter (ATA) designed to bridge the gap between traditional telephone systems and modern Voice over IP (VoIP) networks. As telecommunications technologies continue to evolve, devices like the HT-496 play a crucial role in ensuring seamless communication across diverse platforms.

One of the main features of the HT-496 is its ability to support up to four simultaneous calls. This is particularly advantageous for small to medium-sized businesses that rely on efficient communication to manage customer interactions and internal coordination. The device comes equipped with two FXS ports, enabling users to connect their existing analog telephones directly, ensuring that they can continue using familiar equipment while benefiting from the advanced features of VoIP technology.

The HT-496 supports a variety of voice codecs, including G.711, G.726, G.729, and G.722, allowing for high-quality audio transmission even in bandwidth-constrained environments. This flexibility ensures users can choose the codec that best fits their specific network conditions, optimizing both call clarity and resource efficiency.

In terms of management and security, Grandstream has integrated several advanced technologies into the HT-496. The device includes support for SIP (Session Initiation Protocol), making it compatible with a wide range of VoIP services. Additionally, it features various security mechanisms, such as SRTP (Secure Real-Time Transport Protocol) and TLS (Transport Layer Security), ensuring that voice communications are encrypted and protected from potential threats.

Installation and configuration of the HT-496 are user-friendly, thanks to its web-based interface. This makes it easy for both technical and non-technical users to manage settings, adjust parameters, and monitor system performance. Furthermore, the device supports automatic provisioning, allowing for quick setup with minimal manual intervention.

Another notable characteristic of the HT-496 is its compact design, which enables easy placement in any office environment. Its durable construction ensures reliable operation over time, making it a cost-effective solution for businesses looking to transition to VoIP technology without discarding their existing analog devices.

In summary, the Grandstream Networks HT-496 features a robust design, compatibility with a variety of voice codecs, advanced security protocols, and user-friendly management options. These characteristics make it an essential tool for businesses seeking to enhance their communication systems while maintaining a connection to traditional telephony. By investing in the HT-496, organizations can simplify their transition to VoIP and unlock the full potential of modern telecommunications.
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