Symmetric RTP

 

 

 

Default is No. When set to Yes the device will change the destination to send RTP

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

packets to the source IP address and port of the inbound RTP packet last received by

 

 

 

 

 

 

 

 

the device.

 

 

 

 

 

 

 

 

 

 

 

 

Fax Mode

 

 

 

T.38 (Auto Detect) FoIP by default, or Pass-Through (must use codec PCMU/PCMA)

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Re-Invite after Fax Tone

 

 

 

Default is Enabled. It makes the unit send out the re-INVITE for T.38 or Fax Pass

 

 

 

 

Detection Mode

 

 

 

Through if a fax tone is detected.

 

 

 

 

 

 

 

 

 

 

 

 

Jitter Buffer Type

 

 

 

Select either Fixed or Adaptive based on network conditions. Default is Adaptive.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Jitter Buffer Length

 

 

 

Select Low, Medium or High based on network conditions. Default is Medium.

 

 

 

 

 

 

 

 

∙ High (initial 200ms, min 40ms, max 600ms) Note: not all vocoders can meet

 

 

 

 

 

 

 

 

the high requirement

 

 

 

 

 

 

 

 

∙ Medium (initial 100ms, min 20ms, max 200ms)

 

 

 

 

 

 

 

 

∙ Low (initial 50ms, min 10ms, max 100ms)

 

 

 

 

 

 

 

 

 

 

 

 

SRTP Mode

 

 

 

This option defines different implementation of support SRTP (squired RTP)

 

 

 

 

 

 

 

 

transmission mode. Select between Disabled, Enabled but not Forced, and Enabled

 

 

 

 

 

 

 

 

and Forced. Default is Disabled.

 

 

 

 

 

 

 

 

 

 

 

 

SLIC Setting

 

 

 

Dependent on standard phone type (and location)

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Caller ID Scheme

 

 

 

Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, & NTT Japan

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Polarity Reversal

 

 

 

Default is No. If set to “Yes”, polarity will be reversed upon call establishment and

 

 

 

 

 

 

 

 

termination.

 

 

 

 

 

 

 

 

 

 

 

 

Loop Current

 

 

 

Default is No. Set it to Yes if the traditional PBX you are using with HT70X uses this

 

 

 

 

Disconnect

 

 

 

method for signaling call termination. Method initiates short voltage drop on the line

 

 

 

 

 

 

 

 

when remote (VoIP) side disconnects an active call.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Loop Current

 

 

 

Default value is 200. Here can be configured duration of such voltage drop described in

 

 

 

 

Disconnect Duration

 

 

 

topic above. HT70X supports a Duration Range from 100 to 10000 ms.

 

 

 

 

 

 

 

 

 

 

 

 

Hook Flash Timing

 

 

 

Time period when the cradle is pressed (Hook Flash) to simulate FLASH. To prevent

 

 

 

 

 

 

 

 

unwanted activation of the Flash/Hold and automatic phone ring-back, adjust this time

 

 

 

 

 

 

 

 

value. Default values are 300 minimum and 1100 maximum. HT70X supports a

 

 

 

 

 

 

 

 

range from 40 to 2000 ms.

 

 

 

 

 

 

 

 

 

 

 

 

On Hook Timing

 

 

 

On-hook timing is the minimum time for an on-hook event to be validated. Default

 

 

 

 

 

 

 

 

value is 400 . HT70X supports a range from 40 to 2000 ms.

 

 

 

 

 

 

 

 

 

 

 

 

Gain

 

 

 

Voice path volume adjustment.

 

 

 

 

 

 

 

 

• Rx is a gain level for signals transmitted by FXS

 

 

 

 

 

 

 

 

• Tx is a gain level for signals received by FXS.

 

 

 

 

 

 

 

 

Default = 0dB for both parameters. Loudest volume: +6dB Lowest volume: -6dB.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

User can adjust volume of call on either end using the Rx Gain Level parameter and

 

 

 

 

 

 

 

 

the Tx Gain Level parameter located on the FXS Port Configuration page.

 

 

 

 

 

 

 

 

If call volume is too low when using the FXS port (ie. the ATA is at user site), adjust

 

 

 

 

 

 

 

 

volume using the Rx Gain Level parameter under the FXS Port Configuration page.

 

 

 

 

 

 

 

 

If voice volume is too low at the other end, user may increase the far end volume using

 

 

 

 

 

 

 

 

the Tx Gain Level parameter under the FXS Port Configuration page.

 

 

 

 

 

 

 

 

 

 

 

 

Disable Line Echo

 

 

 

Default is No. If set to “Yes” LEC will be disabled per call base. Recommended for

 

 

 

 

Canceller (LEC)

 

 

 

FAX/Data calls.

 

 

 

 

 

 

 

 

 

 

 

 

Ring Tones

 

 

 

This function lets you configure ring tone cadence preferences. User has 10 choices.

 

 

 

 

 

 

 

 

The configuration, completed in Distinctive Ring Tones block in the same page, applies

 

 

 

 

 

 

 

 

to ring tones cadences configured here.

 

 

Grandstream Networks, Inc.

HT-70X User Manual

Page 29 of 35

 

Firmware Version 1.0.0.18

Last Updated: 03/2012

Page 28 Page 30
Page 29
Image 29
Grandstream Networks HT701 Symmetric RTP, Fax Mode, Re-Invite after Fax Tone, Detection Mode, Jitter Buffer Type, Gain
Page 28 Page 30

HT704, HT701, HT702 specifications

Grandstream Networks has established itself as a powerful player in the telecommunications industry, especially through its Analog Telephone Adapters (ATAs) like the HT702 and HT704 models. These devices are designed specifically for converting analog voice signals into digital data for seamless integration with Voice over Internet Protocol (VoIP) systems.

The Grandstream HT702 is a two-port ATA that allows users to connect two analog phones to a high-speed internet connection. This model is particularly useful for small businesses or residential users looking to integrate legacy phone systems with modern VoIP technology. One of the key features of the HT702 is its support for the SIP (Session Initiation Protocol) standard, ensuring compatibility with a wide range of VoIP providers. Additionally, it supports advanced telephony features like call transfer, call waiting, and three-way calling, enhancing communication efficiency.

The HT704, on the other hand, is a four-port ATA, offering greater flexibility for users needing to connect multiple devices. It shares many of the same features as the HT702, including SIP support and telephony functionalities, but with additional ports, it is better suited for larger environments. Both models come equipped with advanced security mechanisms, such as AES encryption, which safeguards voice communications.

With user-friendly web-based configuration, the HT702 and HT704 allow for easy setup and management, making them accessible even for those without extensive technical knowledge. Moreover, both devices feature auto-provisioning capabilities, which simplify deployment across multiple units, making them ideal for businesses looking to scale their operations.

The HT702 and HT704 are built with high-quality materials, ensuring durability and long-term performance. They also boast low power consumption, making them an energy-efficient choice. Support for high-definition voice codecs enhances audio quality during calls, providing users with crystal-clear communication.

In summary, Grandstream's HT702 and HT704 Analog Telephone Adapters are robust solutions for anyone looking to transition from traditional telephony to a modern VoIP setup. Their advanced features, security standards, and ease of use make them a reliable choice for both home and business users seeking efficient and effective communication solutions.
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