NAT IP phones to PS1 or base server?

[k.parrent]

Installing my first system that includes a PS1. I will have some IP phones that will NAT to a public IP to register to the mivoice 250. Does the public static need to be NAT'ed to the LAN address of the PS1 or base server?

[snowcloned]

You will need to have 2 separate public IP addresses, one with ports forwarded to the Base chassis, and one with ports forwarded to the PS-1. The remote IP phones should be programmed to connect back to the public IP address of the Base; but the UDP ports for RTP need to be forwarded on both public IPs.

In DB Programming, you will need to enter the public IP address of the Base under System > Devices and Feature Codes > IP Connections > (P6000)Processor Module > NAT IP Address; put the public IP of the PS-1 under System > IP Settings > System NAT IP Address (changing these fields requires a reboot to take effect)

[Tech Electronics]

k.parrent,

Well I guess I should clear a few things up first and then you can make a better decision on how you need to proceed.

When you say that you have a PS-1 attached to your system then most likely I would also assume that you have a PEC-1 as well for the extra DSP resources; this is not mandatory although a good practice.

So, with a PS-1 added to the system nothing really changes on how remote (teleworker) phones work unless they are SIP phones. The big difference is if you have a PEC-1 added to the system as well and that is where there are differences on how the firewall will need to be setup to support that configuration.

All call data is handled by the PM (Base Processor Module), but the audio can be on either the PM or the PEC-1 and they use different UDP ports for each one. Typically this happens at the same time, viola one-way audio. This means that you would need [2] Public IP Addresses with one pointing to the PM and one pointing to the PEC-1. I am going to assume that you know all the ports needed, but the difference is really between the RTP Audio like this.

RTP Audio towards the PM: 6004-6261 [UDP]
RTP Audio towards the PEC-1: 6604-7039 [UDP]

Now, if you have SIP then that has to go to the PS-1 if there is one equipped otherwise it goes to the PM, but you also have to put the Public IP in System > IP Settings > System NAT IP Address or you will have no audio after the call is made; clear as mud.

May I also suggest that if you have QoS setup on your network that you change your Audio RTP Type of Service and Data Type of Service from 0 [IP Precedence] to 184 [Expedited Forwarding] for the DSCP when going through Differentiated Services.

Hopefully that clears things up a bit for you.

Thanks,

TE

[k.parrent]

Thanks for the advice guys, I appreciate it.
Tech E - I only have the PS1 and no PEC-1. I am currently only using SIP devices on this system and the customer intends to go to remote IP desk phones later on. They are using a bria app on smart phones, so they are coming in from the public internet. The app is registering successfully to the pbx, and I tested a call and I have one way audio. The SIP user is hearing audio but the far end is not. I checked my SIP phone account to make sure it is set for NAT. And I have the public IP set up in IP settings - NAT IP address. Is this probably a router/firewall issue?

[Tech Electronics]

k.parrent,

Well it could be a few things, but the two most obvious would be whether you have the SIP Peer Group setup for NAT not Native. The other would be that the Firewall is performing some sort of SIP Transformation that is not needed for the solution to work.

On the Fortigate this is called SIP ALG or something along those lines and requires a reset before it will work properly.
On the SonicWall this is called SIP Transformations or something along those lines and does not require a reset to work properly.

Other Firewalls could have different terminology and work differently, but I would have them do a reset before giving up on any change.

Thanks,

TE