I installed a 3120 at a customer site after programming and updating firmware to R10-11-0. This is a new install. The customer has a PBX running SIP trunks. Inbound calls consistently lose audio after approximately 30 seconds. I had this same issue at another site running the same configuration, and replaced that 3120 with a different 3120 that is still running 18.02.01.00.E. That seems to have corrected the problem at the second site. Packet captures by the SIP provider show the RTP ports seem to be changing. It has been recommended that I ensure the "ip firewall nat-preserve-source-port" command has been applied. I have looked for this in the GUI, but haven't been able to locate it. Where would this be, and how do I enable it? Has any one else experienced this issue, and if so how did you correct it?
Oddly enough I have had the same issue but only behind a Verizon DSL account. Changing the account type to static took care of the issue. It appeared however something related to SIP ALG.
I know this is old, but this answer may help others...
When I configure the NV3120 with VoIP (every installation) there are 2 things that need to be done to ensure proper VoIP traffic management.
1) Make sure that the SIP ALG is not enabled. This can be found in Data -> Firewall/ACL -> ALG Settings
2) In your VoIP port forwarding rule of the public interface make sure that Port Translation is not enabled. Data -> Security Zones -> Public -> "Your Port Forward Rule" -> Port Translation radial button to Disabled
This works 100% of the time with all of my VoIP clients running through a NV3120.
Hope this helps!
I am having the same issue with a 3448, and I need a resolution asap. What is mentioned in the thread does not work. I sent this also as a support ticket but I am hoping to get a faster response here if someone has a answer.
Here is what I have. I have a SIP Trunk provider at 18.104.22.168 public, and a phone system at 10.11.52.11 on vlan 3. I have a port forward of 5060 from 22.214.171.124 to 10.11.52.11. All calls complete so the forward is working, BUT after 30 seconds into the call the SIP provider sends a SDP(PRAK) message to the phone system and its dropped, so the phone system does not see it and the call is terminated. All other following packets for the call on port 5060 are also dropped (like the OK response for the BUY message) I have seen this before, and I need a solution. I tried to set "ip firewall nat-preserve-source-port record-source-address" and it does not work.I believe this has to do with the PRAK message is not part of the active session so its dropped. SO, I tried to setup SIP Proxy Transparent because that fixed it the last time BUT as I found out you cannot use SIP Proxy Transparent with a static SIP environment because there is NO SIP registration going on. Incoming calls work fine, but the option ping to the SIP provider gets a reject message from the adtran that its not registered and will not pass threw the traffic. Again, this is a STATIC setup, the provider and phone system sends the options and invites blindly without registration because the security is the IP address. I need to either get the firewall to forward the packets regardless if they are in a session or if there is a active bind on the port, OR have Transparent SIP work without registration.
I will post the solution once I know what it is.
I'm sorry after a closer look at the Wireshark of the provider its a PBX issue where its using the local address for the contact instead of the WAN IP. Something to watch for. Thanks!