Send Flash Event

Enable Call Features

Use Bell-style 3-way Conference

Off-hook Auto-Dial

Proxy-Require

Disable Call Waiting

NAT Traversal (STUN)

No Key Entry Timeout

Preferred Vocoder

Voice Frames per TX

modes supported: 1) in audio which means DTMF is combined in audio signal (not very reliable with low-bit-rate codec), 2) via RTP (RFC2833), or 3) via SIP INFO. Multiple selections of DTMF method are supported.

Default is No. If set to yes, flash will be sent as DTMF event.

Default is Yes. Advanced call features and feature codes functions are supported locally.

Enables Bellcore Style 3-way Conference if parameter is set to “Yes”. *23 is disabled.

This parameter automatically configures and dials User ID or extension number upon off-hook. Only the user part of a SIP address needs is entered here. The HT–496 will automatically append the “@” and the host portion of the corresponding SIP address.

SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.

Default is No.

This parameter defines whether the HT–496 NAT traversal mechanism is activated or not. If activated (by choosing “Yes”) and a STUN server is also specified, then the HT–496 performs according to the STUN client specification. Under this mode, the embedded STUN client will detect if and what type of firewall/NAT is used. If the detected NAT is a Full Cone, Restricted Cone, or a Port-Restricted Cone, the HT–496 will use its mapped public IP address and port in all of its SIP and SDP messages.

If the NAT Traversal field is set to “Yes” with no specified STUN server, the HT–496 will periodically (every 20 seconds or so) send a blank UDP packet (with no payload data) to the SIP server to keep the “hole” on the NAT open.

Default is 4 seconds.

The HT–496 supports up to 6 different Vocoder types including G.711 A-/U- lawG.723.1, G.726, G.729A/B, iLBC.

Users can configure Vocoders in a preference list included with the same preference order in SDP message. The first Vocoder in this list is the appropriate option in “Choice 1”. The last Vocoder in this list is the appropriate option in “Choice 6”.

Number of voice frames transmitted in a single packet. User should be aware of the requested packet time (used in SDP message). This parameter is associated with the first vocoder in the Vocoder Preference List or the actual payload type negotiated between the 2 conversation parties at run time.

e.g. if the first vocoder is configured as G.723 and the “Voice Frames per TX” is set to 2, then the “ptime” value in the SDP message of an INVITE request will be 60ms because each G.723 voice frame contains 30ms of audio.

If the configured voice frames per TX exceeds the maximum allowed value, the HT–496 will use and save the maximum allowed value for the corresponding first vocoder choice.

The maximum value for PCM is 10(x10ms) frames;

for G.726, it is 20 (x10ms) frames; .

for G.723, it is 32 (x30ms) frames;

Grandstream Networks, Inc.

HT–496 User Manual

Page 24 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

Page 24
Image 24
Grandstream Networks HT-496 user manual 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.