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<p class="MsoNormal">Paul, that was my thought on the Perimeta.<o:p></o:p></p>
<p class="MsoNormal">So far, we’ve only had two calls that this issue occurred on… but honestly, not sure how many have 30+ minutes calls on their softphone.<o:p></o:p></p>
<p class="MsoNormal">I just wonder if this was kind of a one-off issue with a specific Verizon cell?<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><b>From:</b> VoiceOps <voiceops-bounces@voiceops.org> <b>On Behalf Of
</b>Paul Timmins<br>
<b>Sent:</b> Thursday, June 10, 2021 11:12 AM<br>
<b>To:</b> voiceops@voiceops.org<br>
<b>Subject:</b> Re: [VoiceOps] "Timeout" on VoIP call traversing Verizon data<o:p></o:p></p>
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<p class="MsoNormal">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.<o:p></o:p></p>
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<p class="MsoNormal"><a href="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">https://community.metaswitch.com/support/solutions/articles/76000007855-product-advisory-perimeta-and-sip-application-level-gateways-algs</a>-<o:p></o:p></p>
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<p class="MsoNormal">On 6/10/21 9:18 AM, Mark Wiles wrote:<o:p></o:p></p>
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<p class="MsoNormal">Hi Dovid,<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">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?<o:p></o:p></p>
<p class="MsoNormal">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)<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">Are you saying that you’re using Metaswitch MaX UC and you’re doing a SIP OPTIONS message every 49 seconds?<o:p></o:p></p>
<p class="MsoNormal">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.<o:p></o:p></p>
<p class="MsoNormal">I’ll bounce your thoughts off of them.<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">Thanks!<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">Mark<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
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<p class="MsoNormal"><b>From:</b> Dovid Bender <a href="mailto:dovid@telecurve.com">
<dovid@telecurve.com></a> <br>
<b>Sent:</b> Thursday, June 10, 2021 8:47 AM<br>
<b>To:</b> Mark Wiles <a href="mailto:mwiles@akabis.com"><mwiles@akabis.com></a><br>
<b>Cc:</b> <a href="mailto:voiceops@voiceops.org">voiceops@voiceops.org</a><br>
<b>Subject:</b> Re: [VoiceOps] "Timeout" on VoIP call traversing Verizon data<o:p></o:p></p>
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<p class="MsoNormal"> <o:p></o:p></p>
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<p class="MsoNormal">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.<o:p></o:p></p>
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<p class="MsoNormal">On Wed, Jun 9, 2021 at 3:27 PM Mark Wiles <<a href="mailto:mwiles@akabis.com">mwiles@akabis.com</a>> wrote:<o:p></o:p></p>
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<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">If there’s a Verizon cellular data guru monitoring here, I’d love to get your insight!<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Otherwise, let me toss this out to the group for thoughts and opinions please…<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">We’re a Metaswitch shop, and use their MaX UC mobile softphone client (iPhone/Android).<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">We had a customer using the MaX UC client on a long call… they were using Verizon cellular data (confirmed by IP address).<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">At thirty (30) minutes into the call, the call “dropped”. The call was re-established, and again, after thirty minutes, the call dropped.<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">We’re pretty sure the user was in a static position (non-mobile)… and logically
<u>assume</u> they were on the same cell tower for both calls that dropped (the Verizon IP was the same).<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">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
<a href="http://1.2.3.4:9876" target="_blank">1.2.3.4:9876</a>.<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">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.<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Metaswitch also looked at the bug/crash logs on the softphone, and confirmed neither was the case.<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">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? <span style="font-family:"Segoe UI Emoji",sans-serif">
😊</span><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Has anyone had experience with such an occurrence… or any thoughts?<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Thank you!<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Mark<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
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<p class="MsoNormal">_______________________________________________<br>
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</blockquote>
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<p class="MsoNormal"><br>
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<pre>_______________________________________________<o:p></o:p></pre>
<pre>VoiceOps mailing list<o:p></o:p></pre>
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<p><o:p> </o:p></p>
<pre>-- <o:p></o:p></pre>
<pre>Paul Timmins<o:p></o:p></pre>
<pre>Clear Rate Communications<o:p></o:p></pre>
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