SIP transport

NAT Traversal (STUN)

SIP User ID

Authenticate ID

work and ONLY outbound proxy can correct the problem.

User can select UDP or TCP or TLS.

This parameter defines whether or not the HT–502 NAT traversal mechanism is activated. If activated (by choosing “Yes”) and a STUN server is also specified, then the HT–502 performs according to the STUN client specification. Using this mode, the embedded STUN client will detect if and what type of firewall/NAT is being used. If the detected NAT is a Full Cone, Restricted Cone, or a Port-Restricted Cone, the HT–502 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–502 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.

User account information, provided by VoIP service provider (ITSP). Usually in the form of digit similar to phone number or actually a phone number.

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

 

Authenticate

 

SIP service subscriber’s account password.

 

 

 

 

Password

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Name

 

SIP service subscriber’s name for Caller ID display.

 

 

 

 

 

 

 

 

 

Use DNS SRV

 

Default is No. If set to “Yes” the client will use DNS SRV to look up server.

 

 

 

 

 

 

 

 

User ID is Phone

 

If the HT–502 has an assigned PSTN telephone number, this field should be

 

 

 

Number

 

set to “Yes”. Otherwise, set it to “No”.

 

 

 

 

 

 

If “Yes” is set, a “user=phone” parameter will be attached to the “From” header

 

 

 

 

 

in SIP request.

 

 

 

 

 

 

 

 

 

 

SIP Registration

 

Controls whether the HT–502 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 w/o

 

Default is No.

If set to “Yes,” user can place outgoing calls even when not

 

Registration

 

registered (if allowed by ITSP) but is unable to receive incoming calls.

 

 

 

 

 

 

 

 

Register Expiration

 

This parameter allows the user to specify the time frequency (in minutes) the

 

 

 

 

 

HT–502 refreshes its registration with the specified registrar. The default

 

 

 

 

 

interval is 60 minutes (or 1 hour). The maximum interval is 65535 minutes

 

 

 

 

 

(about 45 days).

 

 

 

 

 

 

 

 

 

Local SIP port

 

Defines the local SIP port the HT–502 will listen and transmit. The default

 

 

 

 

 

value for FXS port 1 is 5060. The default value for FXS port 2 is 5062.

 

 

 

 

 

 

 

 

Local RTP port

 

Defines the local RTP-RTCP port pair the HT–502 will listen and transmit. It is

 

 

 

 

 

the base RTP port for channel 0. When configured,

 

 

 

 

 

 

channel 0 uses this port _value for RTP and the port_value+1 for its RTCP;

 

 

 

 

 

channel 1 uses port_value+2 for RTP and port_value+3 for its RTCP.

 

 

 

 

 

The default value for FXS port 1 is 5004. The default value for FXS port 2 is

 

 

 

 

 

5012.

 

 

 

 

 

 

 

 

 

 

Use Random Port

 

This parameter forces the random generation of both the local SIP and RTP

 

 

 

 

 

ports when set to Yes. This is usually necessary when multiple HT–502 are

 

 

 

 

 

behind the same NAT.

 

 

 

 

 

 

 

 

 

 

 

Grandstream Networks, Inc.

 

 

HT-502 User Manual

Page 24 of 35

 

 

 

 

Firmware 1.0.0.39

Last Updated: 03/2007

Page 24
Image 24
Grandstream Networks HT-502 Authenticate, Password Name, Use DNS SRV, User ID is Phone, Number, SIP Registration