[VoiceOps] "Timeout" on VoIP call traversing Verizon data

Thu Jun 10 14:30:35 EDT 2021

Since I’m as dumb as a bag of hammers when it comes to cellular data… what do you think the NAT timeout might standardly be, before the pin-hole goes away?

Strange we’ve not run into this before.

From: VoiceOps <voiceops-bounces at voiceops.org> On Behalf Of Paul Timmins
Sent: Thursday, June 10, 2021 11:12 AM
To: voiceops at voiceops.org
Subject: Re: [VoiceOps] "Timeout" on VoIP call traversing Verizon data

The perimeta should auto-detect the NAT and start a "fast register" in their parlance. You might want to look into this and possibly force nat on your MaXUC instead of using nat autodetect, and make sure fast register is configured. It will handle keeping the signaling portion open for you.

https://community.metaswitch.com/support/solutions/articles/76000007855-product-advisory-perimeta-and-sip-application-level-gateways-algs<https://linkprotect.cudasvc.com/url?a=https%3a%2f%2fcommunity.metaswitch.com%2fsupport%2fsolutions%2farticles%2f76000007855-product-advisory-perimeta-and-sip-application-level-gateways-algs&c=E,1,y0GVFh4InJbiGA_p7LsIOQ53DOKFu20uC7tzS2O0aprTQyEoSEYfcuCoPnEPgoDIKx6FG9ffmXeQPB5RVGEhzUKd-y_ZSrmXJR4DmkX-uw,,&typo=1>-

On 6/10/21 9:18 AM, Mark Wiles wrote:
Hi Dovid,

So just thinking about this… granted, there wasn’t SIP traffic for “X” amount of time… but there would have been RTP… so wouldn’t that have been seen as traffic?
Hmmm… but as soon as I typed that, SIP traffic’s on one port… RTP traffic’s on another port… so even with the RTP flowing along and happy… the SIP’s another matter… right?  Duh!  (I’ve not had my coffee yet)

Are you saying that you’re using Metaswitch MaX UC and you’re doing a SIP OPTIONS message every 49 seconds?
I totally agree it does sound like a NAT pinhole is closing.  It would seem that if that’s the case, Meta would have run into this before and had “recommendations” to address this.
I’ll bounce your thoughts off of them.

Thanks!

Mark

From: Dovid Bender <dovid at telecurve.com><mailto:dovid at telecurve.com>
Sent: Thursday, June 10, 2021 8:47 AM
To: Mark Wiles <mwiles at akabis.com><mailto:mwiles at akabis.com>
Cc: voiceops at voiceops.org<mailto:voiceops at voiceops.org>
Subject: Re: [VoiceOps] "Timeout" on VoIP call traversing Verizon data

If I had to guess Verizon is using CGNAT and since there is no traffic for X amount of time the NAT hole for the SIP traffic is closed. When you send a re-invite at the 30 minute mark that session as far as Verizon's CGNAT devices are concerned have been closed a long time ago. You would need to send a packet to the phone or have the phone send to your switch some sort of traffic (we send SIP OPTIONS every 49 seconds) to ensure that the session stays alive.

On Wed, Jun 9, 2021 at 3:27 PM Mark Wiles <mwiles at akabis.com<mailto:mwiles at akabis.com>> wrote:
If there’s a Verizon cellular data guru monitoring here, I’d love to get your insight!

Otherwise, let me toss this out to the group for thoughts and opinions please…

We’re a Metaswitch shop, and use their MaX UC mobile softphone client (iPhone/Android).

We had a customer using the MaX UC client on a long call… they were using Verizon cellular data (confirmed by IP address).
At thirty (30) minutes into the call, the call “dropped”.  The call was re-established, and again, after thirty minutes, the call dropped.
We’re pretty sure the user was in a static position (non-mobile)… and logically assume they were on the same cell tower for both calls that dropped (the Verizon IP was the same).

Looking at Metaswitch SAS (their diagnostics tool), at the thirty minute mark, we send out a re-INVITE message to the softphone client… and we receive no reply… so after ten seconds, we breakdown the call assuming they’re gone.  Then about eight seconds later, we see an INVITE message from the softphone’s same IP address (with the same Call ID)… however, it’s coming from a different port.  So to be clear, the original call setup and connection was using 1.2.3.4:6789… then eight seconds after we ended the call with a BYE (assuming they were gone due to lack of reply), we get an INVITE (with the same Call ID) from 1.2.3.4:9876<http://1.2.3.4:9876>.

Metaswitch looked at the diags from the softphone (we downloaded them), and they’re confirming that the softphone never received our re-INVITE at the 30 minute mark.

Metaswitch also looked at the bug/crash logs on the softphone, and confirmed neither was the case.

It almost sounds like a NAT thing going on… but I’m pretty ignorant when it comes to cellular data.  It looks to me as if the Verizon side simply changed port numbers, and assumed we’d know maybe via mental telepathy?  😊

Has anyone had experience with such an occurrence… or any thoughts?

Thank you!

Mark

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