Quote:
Originally Posted by vientito
Besides resorting to a non-standard SIP port, do i need to do something special about STUN server setting or RTF port range at all?
|
If you are using NAT_keep_alive and STUN, and not using port forwarding in the router then you can leave the SIP and RTP ports set at factory defaults on both ATA units. Your NAT router will sort out the SIP and RTP packets appropriately. Settings would be as follows:
(under SIP tab)
Handle_VIA_received: yes
Handle_VIA_rport: yes
Insert_VIA_received: yes
Insert_VIA_rport: yes
Substitute_VIA_Addr: yes
Send_Resp_To_Src_Port: yes
STUN_Enable: yes
STUN_Test_Enable: yes
STUN_Server: stun.voxalot.com:3478 (or any STUN server such as 'stun01.sipphone.com:3478' or 'stun.sipgate.net:10000')
NAT_Keep_Alive_Intvl: 179
(under Line_1 and Line_2 (or PSTN_Line) tabs)
NAT_Mapping_Enable: yes
NAT_Keep_Alive_Enable: yes
NAT_Keep_Alive_Msg: (this entry should be completely blank)
NAT_Keep_Alive_Dest: $PROXY
Register Expires: 3600
========================
On the other hand, if you want your phone service to continue no matter if the stun server is working then you should configure without stun. You will need to use unique SIP ports and RTP port range for each ATA, and configure as above except for the following changes:
1. Forward the appropriate SIP ports and the RTP port ranges from the router to each ATA.
2. Set "STUN Enable:" no
3. Set "NAT Keep Alive Enable:" no
4. In order for the ATAs to know their public IP address, make sure each ATA is registered with at least one SIP registrar, or enter your public IP address into the "EXT IP:" field).