Grandstream Networks HT704, HT702 SIP transport, NAT Traversal Stun, SIP User ID, Authenticate ID

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environments. If symmetric NAT is detected, STUN will not work and ONLY outbound

 

 

 

 

 

proxy can correct the problem.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

SIP transport

 

User can select UDP or TCP or TLS. Default is UDP.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

NAT Traversal (STUN)

 

This parameter defines whether or not the HT70X NAT traversal mechanism is

 

 

 

 

 

activated. If activated (by choosing “Yes”) and a STUN server is also specified, then the

 

 

 

 

 

HT70X performs according to the STUN client specification. Using this mode, the

 

 

 

 

 

embedded STUN client will detect if and what type of firewall/NAT. If the detected NAT

 

 

 

 

 

is a Full Cone, Restricted Cone, or a Port-Restricted Cone, the HT70X 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 HT70X 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.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

SIP User ID

 

User account information, provided by VoIP service provider (ITSP). Usually in the form

 

 

 

 

 

of digit similar to phone number or actually a phone number. HT701 and HT702 only

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Authenticate ID

 

SIP service subscriber’s Authenticate ID used for authentication. Can be identical to or

 

 

 

 

 

different from SIP User ID. HT701 and HT702 only

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Authenticate Password

 

SIP service subscriber’s account password. HT701 and HT702 only

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Name

 

SIP service subscriber’s name for Caller ID display. HT701 and HT702 only

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

DNS Mode

 

One from the 3 modes are available for “DNS Mode” configuration:

 

 

 

 

 

 

-A Record (for resolving IP Address of target according to domain name)

 

 

 

 

 

-SRV (DNS SRV resource records indicates how to find services for various protocols)

 

 

 

 

 

-NAPTR/SRV (Naming Authority Pointer according to RFC 2915)

 

 

 

 

 

 

One mode can be chosen for the client to look up server.

 

 

 

 

 

 

The default value is “A Record”

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Tel URI

 

The default setting is “Disabled”. If the phone has an assigned PSTN

 

 

 

 

 

 

Number, this field should be set to “User=Phone” then a

 

 

 

 

 

 

“User=Phone” parameter will be attached to the “From header” in the SIP

 

 

 

 

 

request to indicate the E.164 number. If server supports TEL URI format, then this

 

 

 

 

 

option needs to be selected.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

SIP Registration

 

Controls whether the HT701 needs to send REGISTER messages to the proxy server.

 

 

 

 

 

The default setting is Yes.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Unregister on Reboot

 

Default is No.

If set to Yes, the SIP user’s registration information will be cleared on

 

 

 

 

 

reboot.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Outgoing Call without

 

Default is No. If set to “Yes,” user can place outgoing calls even when not registered (if

 

 

 

Registration

 

allowed by Internet Telephone Service Provider) but is unable to receive incoming

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

FIRMWARE VERSION 1.0.3.1

HT70X USER MANUAL

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Contents Grandstream Networks, Inc Index Software Upgrade HT70X Connection Diagram GNU GPL Information Change LOG Safety Compliances WarrantyWelcome Firmware Version Connecting the HT70X Connect Your HT70XEquipment Packaging Reset Power LEDInternet LED Link/Activity LEDPhone LED LED-13 LED-14LED-15 LED-16Software Features Overview HT70X FeatruesIP Signaling FeaturesDhcp Server/Client Telnet ServerHardware Specification EMCReset the HT for the new IP address to take Effect Basic OperationsUnderstanding HT70X Voice Prompt Main MenuSee Restore Factory Default Setting section See Make a Direct IP CallPlacing a Phone Call Phone or Extension NumbersCall Hold Call WaitingCall Transfer WAY Conferencing Attended TransferFAX Support Call Features Enable Srtp Disable SrtpBlind Transfer Flash/Hook Configuration Guide Configuring the HT70X Through Voice PromptsConfiguring the HT70X VIA WEB Browser Important Settings Access the WEB Configuration MenuEnd User Password Web PortPPPoE password PPPoE Service NameIP Address Dhcp hostnameReset Type MTZ+6MDT+5Allow Dhcp server to Set Time Zone LanguageForward Busy Delayed NATDND FXSAdmin Password Firmware UpgradeLayer 3 QoS Layer 2 QoSHTTP/HTTPS ACS URL Firmware VersionInternet Telephone Service Provider Syslog Level Send SIP LogPrimary Radius Primary Radius AuthRadius Retry Download DeviceAuthenticate ID Authenticate PasswordDNS Mode Unregister on RebootEnable Ring-Transfer Disable Bellcore StyleRegister Expiration Registration Retry WaitSIP T1 Timeout Disable DtmfEnable Call Features Delay Proxy-RequireDisable Call Waiting Disable Call-WaitingDisable Receiver Disable Reminder RingDial Plan Prefix Use # as Dial KeyDial Plan Dial Plan Rules Special Feature Caller Request TimerCallee Request Timer Force TimerVAD Slic Setting Enable Hook FlashDisable Line Echo Canceller LECEnable Ports SIP Use IDProfile ID Hunting GroupSaving the Configuration Changes Rebooting the HT70X from RemoteConfiguration Through a Central Server Port Sip port that will be annexed to the IP address aboveFirmware Version Software Upgrade Firmware Upgrade Through TFTP/HTTP/HTTPSInstructions for Local Firmware Upgrade Using Tftp Server Instructions for Upload from Local DirectoryConfiguration File Download Firmware and Configuration File Prefix and PostfixManaging Firmware and Configuration File Download Firmware Version Restore Factory Default Setting Factory ResetReset from web interface Reset Type
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HT704, 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.