BAsic-but importatnt questions...

[hunok99]

I've been dealing with this Voip, PAP, SPA, GXP2000 for over a year and thanx to some of you ( Howard) my stuff is working OK.
I still do not get some of the most important things under the settings, like why to use and when to use YES or NO on the following strings:

NAT Support Parameters
Handle VIA received:yes no
Handle VIA rport:yes no
Insert VIA received:yes no
Insert VIA rport:yes no
Substitute VIA Addr:yes no
Send Resp To Src Port:yes no
STUN Enable:yes no
STUN Test Enable:yes no

Use Outbound Proxy:yes no
Use OB Proxy In Dialog:yes no
Register:yes no
Make Call Without Reg:yes no
Ans Call Without Reg:yes no

[voip_addict]

Hunok99,

From http://www.voip-info.org/wiki/view/S...umentation+SIP


NAT Support Parameters

Handle VIA received: Indicates whether the SPA should react when it encounters a "Via: ...;received=someIPaddr". If set to "yes", then the SPA will use the IP address specified in the "received=" as the public IP address of the SPA.

Handle VIA rport: Indicates whether the SPA should react when it encounters a "Via: ...;rport=someUDPport". If set to "yes", then the SPA will use the UDP port specified in the "rport=" as the public UDP port of the SPA.

Insert VIA received: Indicates whether the SPA should add the SIP header, "Via: ...;received=someIPaddr". If set to "yes", then SIP replies from the SPA will add a "Via:" header that reflects the public IP address the SPA saw originating the SIP request.

Insert VIA rport : Indicates whether the SPA should add the SIP header, "Via: ...;rport=someUDPport". If set to "yes", then SIP replies from SPA will add a "Via:" header that reflects the public UDP port the SPA saw originating the SIP request.

Substitute VIA Addr : Indicates whether the public IP address and port discovered by STUN should be used instead of the SPA's current IP address and port (which may be a private address).

Send Resp To Src Port : Indicates whether to ignore the "Contact:" header and instead send the reply directly back to the IP address and port originating the SIP message.

STUN Enable: Indictates whether to use Simple Traversal of UDP through NAT STUN.

STUN Test Enable: Indictates whether the SPA should perform STUN tests to determine the type of NAT the SPA is behind. The results of the test will show up in the SIP headers of a REGISTER. You will see: Warning: 399 spa "Some Response". The different Responses can be: "Unknown NAT Type", "STUN Server Not Reachable", "STUN Server Not Responding", "Open Internet Detected", "Symmetric Firewall Detected", "Full Cone NAT Detected", "Restricted Cone NAT Detected", or "Symmetric NAT Detected". Additionally, the results of the test influence the behavior of other NAT-related services on the SPA (i.e. Nat Mapping might be automatically disabled).

STUN Server : The STUN server to use. This could be a domain which will be used for a DNS SRV up for a _stun._udp service, a hostnameort, or IPaddrort.

EXT IP : The public IP address to use for replacement of private addresses in the "Via:", "Contact:", and "c=" line in the SDP.

EXT RTP Port Min : The minimal RTP audio port to use for replacement of a private UDP port specified in the "m=" line in the SDP.

NAT Keep Alive Intvl : Maximum duration (in seconds) between two packets used to keep open the pinhole created in the NAT/firewall.


This part I'll answer on my own to the best of my knowlege...

Use Outbound Proxy:yes no
If set to 'yes' call will be routred out over the proxy you have configured.
(basically passing through your providers sip proxy)

Use OB Proxy In Dialog:yes no
This is an area where I can learn a little myself. I know it has to do with additional info being added to the headers. Can someone please give some more detail on this?

Register:yes no
Some providers required your ATA to be "registered" in order to use thier
service. If your provider requires registration, this setting must be set to "yes" in order to make calls. Also, if you receive incomming calls from your provider this setting will have to be set to "yes" regardless, so the provider knows where your ATA is and can send the call to the appropiate place.

Make Call Without Reg:yes no
If set to 'no' your ATA will attempt to place a call even if it is not registered to your provider. Keep in mind that if your provider requires registration and your ata is not registered or registration fails you will not be able to make calls, this includes calls, dialed out on gateway accounts configured in your dial plan.

Ans Call Without Reg:yes no
If set to 'no' and your ATA loses registration, you will not be able to receive calls. Also if your making calls peer to peer without a provider this setting will need to be set to 'yes' because in this case calls are not taking place through a registered provider. I leave this setting set to 'no' as I receive direct peer to peer calls and inbound calls from a DID that points directly to my ATA.

If there's anything missing here or someone wants to get into the nity gritty about registration and the information thats added to headers when 'use outbound proxy in dialog - yes' is selected, please do so.



VoIP_Addict

[hunok99]

Thanks for your time!!!