Ok, Who knows what's going here...Here is my Scenario..
We're looking at a new Phone Platform and I'm only able to get a NAT to work part of the time. First, when the IP Phone loads, internal address of 172.23.1.1, It connects out to the Platform IP of 126.96.36.199, downloads the config from it's TFTP Service. Since we don't want our Voice traffic mingled in with our other public traffic, we've NAT'd 172.23.0.0 /16 to a Public IP of 188.8.131.52, the NAT Works perfectly here. Next, once the TFTP Load is complete, the Phone tries to register via SIP, same Internal Address 172.23.1.1 to the platform of 184.108.40.206, however the same NAT Statement is not being used. I've ran numerous PCAP's, changed NAT several times, moved rules, but everything appears correct. Furthermore, when I run test nat-policy-match with the proper destination and source on port 5060 and protocol 17, it test out correctly.
Inside Zone to Destination Zone, source address 172.23.0.0/16 to destination address 220.127.116.11, any interface and any service translate type Dynamic IP address of 18.104.22.168. Anyone have any idea what's going on and why the NAT isn't getting applied 100% of the time? When I look at the Web GUI of the phone platform, it's showing my phone as being registered with our Public IP of 22.214.171.124. I am able to place and receive a call, however there is not audio. Running another PCAP for a call session, all UDP packets from 172.23.1.1 are getting dropped.
One thing to note, SIP ALG is turned enabled, though I'm not sure if that's the issue. Do try to get around SIP ALG, I created a custom Application, mirrored from SIP, and setup an Application Override with Inside/Outside, all IP's and UDP 5060, but still having the issue.
I've opened a Support Case, but with it being a low priority, thought I'd reach out to the community to see if anyone has ran into this issue inthe past.
We're running 8.0.8 PAN-OS version.
with dynamic IP you may be running out of ip-port combinations and breaking NAT because you only have 1 single IP and are forcing port reuse (oversubscription of more than 8x will be reached quickly)
if your SIP implementation doesn;t allow the varying source ports, you'll need a bigger subnet to NAT your phones
I've changed to dynamin-ip-and-port, however I'm still getting the same results. Currently, we're just testing with one phone back to the new PBX solution. Support hasn't been any help, they've asked that I clear all sessions, would be suprised if that actually works. Doesn't matter what I try, the TFTP Download from the PBX utilizes the NAT however the SIP Registration does not, same source and Destination addresses for both the TFTP Download and SIP Registration.
Have you set a filter and checked the global counters? SIP has this thing where it sometimes reuses all the session parameters (source ports et al) of the previous session, which the firewall doesn't like
that will likely show up in error counters
additionally, you could try disabling the ALG on the sip application, or try an app override altogether
We aren't getting any audio, either way. My Phone Rings just as it should, when I answer there isn't any audio either way. Running PCAP's, I keep seeing 401 unauthorized responses back from the PBX.
Even wierder, just called that test phone from my cell phone, I just left the call established and after about 4 Minutes, i had audio both ways?
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