IP Office v11 drops Flowroute SIP audio after 15 minutes

[ZL_OR]

After working with 2 different Avaya Support Companies we still have this issue. On Incoming and Outgoing calls the audio for both ways gets lost at almost exactly 15 minutes, it doesn't matter what kind of number we are calling. This happens 99 percent of the time, the few times it hasn't happened it drops at 30 minutes. Flowroute suggests that its an RTP issue which we have changed those settings around without success. We have port forwarding to the IP office. I have screenshotted all of the relevant pages of the manager, unifi controller, and email from Flowroute. The route is Centurylink Fiber > Unifi USG Pro 4 > Unifi Switch > IP Office. We have 2 channels from flowroute. We primarily use J179s.

Screenshots

https://photos.app.goo.gl/sPfmKgJVtDnyCtZ79

Please share your thoughts. If I missed any information please let me know. I have spent many hours searching the web with no luck.

Thanks,

Zach

 

[TouchToneTommy]

I see you're not using Network Topology - with Flowroute we normally use LAN-1, put the public IP address in manually, set the ports to 5060/5060/5061, and clear the STUN address. Make sure ALG and Inspections are turned off on the firewall

 

[ZL_OR]

Can you confirm that this is correct? I put in my public IP address for the fiber. If the fiber is down I have cable which the USG will failover to. Will that be in issue?

https://photos.app.goo.gl/xcZCLujSDhTcTEEv7

It will not let me run the STUN its greyed out. I saved which required restart then attempted to run the stun and no address came up.
Thank you for the quick reply TouchToneTommy

-Zach

 

[TouchToneTommy]

Don't know why that is greyed out - is it a live configuration? You don't need to run stun, you've told it what your public IP is, and the IPO will construct the messages to your SIP provider with that IP address.

 

[_COYS_]

I know this isn't super helpful, but we had a very similar issue with one site on flowroute. Calls would disconnect after 15 minutes.

After going back and forth making changes here and there, we ended up porting the customer to another provider.

With that being said, we use Flowroute for a few other sites and have never had this problem.

 

[derfloh]

Perhaps a Reinvite after 15min with wrong SDP information? Did you check any traces so far?

Need some help with IP Office?

https://www.fwilke.com/home

CLI based cale blocking:

https://blog.fwilke.com/ip-office-block-calls-from-specific-cli/

SCN fallback over PSTN:

https://blog.fwilke.com/ip-office-scn-fallback/

 

[ZL_OR]

TouchToneTommy - When I switch the Firewall NAT type to the unknown like you did I lose connection to flowroute. I did try and run the STUN and it changed my Public IP to 0.0.0.0. When I put the Public IP back in switch it back to Open Internet it works fine.(I left all other settings as you recommended) Could this be part of my issue?

_COYS_ - Thank you for your input. Which provider did you switch them to and get better results with similar cost model. (With flowroute we pay $20 dollars/month for 2 channels, 600 Calls and $1400 Minutes)

derfloh - I'm new to SIP and IP Office where should I obtain SDP information? Flowroute? Are you suggesting I do a tracert to the flowroute IP? I have looked at the monitor log when it cuts out but nothing pops out to me.

 

[derfloh]

Hard to describe.

With SIP filters only in Monitor you will see all SIP messages. Some messages (Invite, OK and sometimes others) contain a SDP part that shows the available codecs as well as port and address information.

You should check if you get an Invite message after 15 mins and compare the SDP information with those of the initial call initiation.

Need some help with IP Office?

https://www.fwilke.com/home

CLI based cale blocking:

https://blog.fwilke.com/ip-office-block-calls-from-specific-cli/

SCN fallback over PSTN:

https://blog.fwilke.com/ip-office-scn-fallback/

 

[TouchToneTommy]

Open Internet and Full Cone NAT are other 2 usual choices, depending on the firewall in use. SonicWall uses Unknown.

· Open Internet
No action required. If this mode is selected, settings obtained by STUN lookups are ignored. The IP address used is that of the system LAN interface.

· Full Cone NAT
A full cone NAT is one where all requests from the same internal IP address and port are mapped to the same external IP address and port. Furthermore, any external host can send a packet to the internal host, by sending a packet to the mapped external address. SIP packets need to be mapped to NAT address and Port; any Host in the internet can call in on the open port, that is the local info in the SDP will apply to multiple ITSP Hosts. No warning will be displayed for this type of NAT because the system has sufficient information to make the connection).


You might also want to google unifi usg pro 4 udp timeout

 

[bahmonkeys]

This is a Firewall issue not an IPO issue. Monitor the SIP Trunks in Monitor tool check for TX and RX opts. Make sure the there is a rule in your firewall to the SIP provider.

Rule: Source Address: IPO Address
Destination Address: SIP Provider
Port: 5060 (usually)
Protocol: UDP (usually)

 

[ZL_OR]

TouchToneTommy - I have been looking around the udp unifi issue thank you for clueing me in to that.

Bahmonkeys - I am convinced it is a firewall/router issue now too. I have attached 2 screenshots (not at my desk) since there is multiple Flowroute IPs hosted on the Amazon Cloud how do I configure it. Do I use the domain name, or add all the Flowroute IPs, or can I put in the /28 ipaddress for us-west. I have never had to do static routes before. There is no option for udt on static routes only on Port Forwarding which I have in place for Port 5060 udp to ip office. Please advise on what IP goes where.

Thank you

 

[DavidCT]

Flowroute requires that you have the RTP port range configured in the IP Office open from ANY to the private IP of the PBX on your firewall. Have you done that?

 

[ZL_OR]

DavidCT - I believe I did that correctly. If you could look at the link in my original post and see if I did the rtp port forward correct?

 

[DavidCT]

You don't have it correct.
You are only allowing audio from one of the Flowroute POPs, not "anywhere".

 

[ZL_OR]

Thank you I switched it to anywhere. Made another call and at 15 minutes it stays connected but audio in both directions cuts out like it has been doing.

 

[DavidCT]

Have you tried:

1) On the Network Topology Tab, changing the value for binding refresh?
2) A different firewall make/model?
3) Changing your FlowRoute POP to a different server, even if it's further away from your location?

 

[aarondyck]

If I had to guess I Would say that your session timer is not negotiating properly. Play with the Session Timers section of your SIP Line. The default is Re-Invite with the timer set to On Demand. It doesn't sound like Flowroute is demanding it, and so your call is being disconnected. By adjusting the timer to a value lower than the 900 seconds you are currently experiencing you may be able to resolve your issue.

 

[ZL_OR]

I think we have a winner

Thank you all for taking the time to help me with this issue.

 

[ZL_OR]

I Spoke too soon - I think we are making progress but it still drops 50% of the time at 15 minutes

 

[derfloh]

The question is... Who quits the call? IPO or the carrier? And what messages do you see at that moment?

Need some help with IP Office?

https://www.fwilke.com/home

CLI based cale blocking:

https://blog.fwilke.com/ip-office-block-calls-from-specific-cli/

SCN fallback over PSTN:

https://blog.fwilke.com/ip-office-scn-fallback/