GS Tutorials - UCM: SIP NAT Settings

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GS Tutorials - UCM: SIP NAT Settings
In this tutorial, we will be discussing the UCM SIP NAT settings and how to use it to your advantage.
There will be certain deployments where the UCM is all setup with users registered, but there's just one thing, your inbound calls are not reaching it's destination or a more common issue is one way audio.
The settings that we will be talking about today:
External Host
Use IP address in SDP
External TCP Port
Local Network Address
You can find more details on these options from our user manual here:
www.grandstream.com/sites/defa...
Interesting in reading more about Grandstream's IPPBX series? Please visit our UCM product page for more info:
www.grandstream.com/products/i...

[@drostoker]

I see you are on a roll. Good topic to cover as it is a popular gotcha. Keep them coming!

[@ronaldnexus2378]

Just bought a Grandstream UCM6202. This was a perfect video, been trying to fix this issue for several days, now it works! Cool beans!

[@alanavella]

Hello, I just wanted to thank you I was having trouble on and off regarding audio with my remote extensions. The first portion of the video on the public IP solved the issue. Liking grandstream products more and more. Thanks again, and keep up these videos!

[@VoIPPortland]

ALWAYS change the web login away from 8089. 8089 gets hammered 24/7/365 within hours of plugging in any UCM with Port forwarding. Super easy, just need to add the port extension on your external login such as 82.32.141.12:8002 - or whatever 4-digit port number your choose. Under network, HTTPS settings.

[@chucksw1]

Perfect, Im getting dropped calls around 30 sec, I will give this a try!! Thanks!

[@GrandstreamNetworks]

This sounds like you may have a network problem. If a provider does not receive an ACK from the UCM in a timely manner it will drop the call or if no rtp is being passed. Running a packet capture would help determine where the problem is.

[@datalinq]

one of my customers had this issue and I disabled SDP within the sip settings> NAT tick box. the 30 second drop went away

[@chucksw1]

I did some more testing and found, i get dropped calls after 32 sec when dialing local 7 digit calls using my flowroute trunk, when I use another outbound route with a dial 10 digit pattern and dial the local call with area code and the same flowroute trunk , the call does NOT drop. I also changed the outbound route 7 digit pattern _xxxxxxx to use another trunk provider vitelity outbound and that works good does not drop the call. the outbound route has a privilege level of local. To me this doesnt seem like a nat issue, I took an Ethernet capture of the call drop and I did not see any call issue in wireshark, the call finished with a sip bye. In my case SDP within the sip settings did not make any difference.

[@lthGS]

Chuck, I'd highly recommend opening a support ticket at helpdesk.grandstream.com to have one of our support engineers assist with troubleshooting.

[@datalinq]

Chuck Weinberg did you find a fix for this?

[@arlynweyn9465]

Will video calling work in these setup?

[@axxoaxx288]

my scenario like this im in trouble for inbound call. 1. UCM 6208 is in route mode. SIP trunk is using for making calls . WAN port is terminated with ITSP SIP link. they provided local ip. LAN port is used for IP Phone to register. outbound calls working fine but inbound call fails. no NAT is configured . do i need to configure NAT in this case? i dont know what is the public ip in need to provide

[@kurntechdouglas233]

HAS ANYONE WORK WITH GOIP8 AND THE UCM i can seem to call other networks can someone direct me to get the info on that

[@md.nazmulislammilky8080]

unclear video, can see clearly this video