Optimal NAT settings

[sthostetler]

I am a new reseller and currently have some friends trying out the service. I want to be sure of this before launching it largely. I have a few questions.

I have been reading about NAT and port forwarding with this service. Some say that you need to forward ports, some say the router should handle this.

I agree the router should handle this. But I assume some clients routers are not sufficiently handling this information.

Before I send these units out to real clients I want to be sure that all will work. While primarily I will be on site at installation. I do not want a large amount of issues where customers become uphappy.

So my question is what is the preferred settings before mailing adaptors?

Will using the Voipo provisioning server eliminate some of this headache or not?

Will I manually have to configure the clients router on install to forward these ports?

I have a specific customer that is upcoming that will have two atas behind one router. How do I do this and still use the provisioning? or will I have to manually setup one of these to use another port beside 5060....


My experience with this has been limited, I have a trixbox installation here and never had to forward ports for flowroute. It all just works. I have a router with dd-wrt and never had an issue with this.

THanks for your help!

[genxweb]

for ports

5060-5070 UDP for SIP Registration (only that high if they want to add multiple registrations)

10000-20000 UDP SIP communication ports (actual ports used during conversations)

Voipo offers a stun server which helps with this need: zeus.voipwelcome.com

[sthostetler]

Great support here!!! I really need to be pointed in the right direction when it comes to optimal settings for the adapters.

Thanks

[genxweb]

Here is your answer.

1. Voipo provides a stun server which helps with nating issues. zeus.voipwelcome.com
2. If that dont work here are ports you need:

a. 5060-5061 (at least) UDP / BIDIRECTIONAL: These are SIP registration ports for you to register to the server.

b. 10000-20000 UDP BIDIRECTIONAL: These are the ports SIP communication happens over.

This should resolve nating issues. Also 1 to 1 nat is ideal, if not possible then do port forwarding of all these ports inbound to your ata and allow all out from your ata.

Side note none of my clients that are using Voipo devices have issues with NAT. They plug and play voipo devices are are golden. I believe this is due to zeus stun server. I have about 50 clients using voipo devices the rest are using their sip credentials in pbx systems and softphones.