<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"
"http://www.w3.org/TR/REC-html40/loose.dtd">
<html>
<head>
<meta http-equiv="content-type" content="text/html; charset=utf-8">
<title></title>
</head>
<body style="font-family:Arial;font-size:14px">
<p>Hi Nathan-<br>
<br>
> <span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>In short, I have a hard time believing that MTU issues are the underlying<br>
> cause for many (or even any) VoIP audio delivery problems</span><br>
<br>
Only if it was encapsulated over some form of TCP.<br>
<br>
<span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>> VoIP audio streams other than PCMU-encoded ones, so perhaps it's<br>
> possible other codecs are different</span><br>
<br>
<span style='font-size: 11pt; font-family: "Calibri", "sans-serif";'>It might be worth checking for EVS, which has a lot of SDP options. We've seen some endpoints (handsets) that stop encoding because they didn't understand SDP asks from the receiver. Basically bugs, they get fixed over time, but since EVS is newer and still in adoption phase, that time is stretched out.<br>
<br>
-Jeff</span><br>
<br>
Quoting Nathan Anderson via VoiceOps <<a href="mailto:voiceops@voiceops.org">voiceops@voiceops.org</a>>:</p>
<blockquote style="border-left:2px solid blue;margin-left:2px;padding-left:12px;" type="cite">
<div class="WordSection1">
<p class="MsoNormal"><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Yes, hosts or routers-in-the-middle that don't send ICMP type 3 code 4, or which drop such a message sent by another host instead of forwarding it, do make me upset.</span></p>
<p class="MsoNormal"><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>But...</span></p>
<p class="MsoNormal"><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>In this case we're talking about relatively narrow-band, widely-compressed RTP audio. Admittedly I rarely deal with any VoIP audio streams other than PCMU-encoded ones, so perhaps it's possible other codecs are different (though I'd be surprised...timeliness of delivery in a real-time application like this is far more important than efficiency of packing the data into as few frames as possible), but I personally have never seen an RTP frame that comes close to approaching standard Ethernet MTU. The packets are typically more like a couple hundred bytes large.</span></p>
<p class="MsoNormal"><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>And of course being UDP, TCP MSS doesn't enter into the picture, either.</span></p>
<p class="MsoNormal"><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>In short, I have a hard time believing that MTU issues are the underlying cause for many (or even any) VoIP audio delivery problems...but, as the meme goes, "change my mind"; heh.</span></p>
<p class="MsoNormal"><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>-- Nathan</span></p>
<div>
<div style="border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in">
<p class="MsoNormal"><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b> <span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>VoiceOps [mailto:voiceops-bounces@voiceops.org] <b>On Behalf Of</b> Pinchas Neiman via VoiceOps<br>
<b>Sent:</b> Sunday, March 10, 2024 7:29 AM<br>
<b>To:</b> Alex Balashov<br>
<b>Cc:</b> VoiceOps<br>
<b>Subject:</b> Re: [VoiceOps] One Way Audio - Frontier Comm (Los Angeles area)</span></p>
</div>
</div>
<div>
<div>
<p class="MsoNormal">I have (on a rural area DSL line) a desk phone registered directly on line 1, and line 2 over the VPN, whenever someone on line 1 tells me I couldn't hear you well, I am saying calling you back with another line, every time they will respond immediately Ah. Now your voice is much better.</p>
</div>
<div>
<p class="MsoNormal">TCP connections are also much more reliable over the VPN than direct.</p>
</div>
<div>
<p class="MsoNormal">I am using WG over UDP with MTU 80 bytes lower than the worst case general MTU.</p>
</div>
<div>
<p class="MsoNormal">I digged through my issue, and found that some hops in my long list of local hops (last mile/s) are very unreliable, and not responding when they drop (crime #1) a big packet even if DF was set (crime #2), so best for me was to have wireguard do the fragmentation on my side, as well as signal to the TCP connections to lower their MSS automatically.</p>
</div>
<div>
<p class="MsoNormal">In other cases a VPN will also be able to patch TCP issues related to asymmetric routing, or firewall timeouts.</p>
</div>
<div>
<p class="MsoNormal">To be noted, </p>
</div>
<div>
<p class="MsoNormal">#1 VPN device CPU should be fast enough to do the packaging, as there is usually no hardware assistance for the VPN prepackaging.. a good gigabit router could easily become a source of latency when it involves something more than passing/nating packets between ports</p>
</div>
<div>
<p class="MsoNormal">#2 having a VPN without adjusting the MTU (either manually or automatically) will increase packet loss, this is the source of the myth that VPN is a overhead for VOIP</p>
</div>
<div>
<p class="MsoNormal">My understanding in networking may be flawed but this is my practical experience accumulated so far.</p>
</div>
</div>
<div>
<div>
<p class="MsoNormal">On Sat, Mar 9, 2024 at 4:00 PM Alex Balashov via VoiceOps <<a href="mailto:voiceops@voiceops.org">voiceops@voiceops.org</a>> wrote:</p>
</div>
<blockquote style="border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in">
<p class="MsoNormal">No, it's true, consider me appropriately humbled. I underappreciated the nuance of this issue. I thought we were talking about something closer to the physicality of networks, not packet inspection/filtering/etc.<br>
<br>
-- Alex<br>
<br>
> On 9 Mar 2024, at 08:11, James Cloos <<a href="mailto:cloos@jhcloos.com" target="_blank">cloos@jhcloos.com</a>> wrote:<br>
><br>
>>>>>> "AB" == Alex Balashov writes:<br>
><br>
>>> I don't trust last mile networks to reliably deliver SIP calls. I usually end up putting them into VPNs, TLS, etc.<br>
><br>
> AB> VPNs and TLS make last-mile networks more reliable? :-)<br>
><br>
> on the vpn side, wireguard (which runs over udp) certainly does.<br>
><br>
> I imagine ipsec sometimes can, too. but wg is easier.<br>
><br>
> -JimC<br>
> --<br>
> James Cloos <<a href="mailto:cloos@jhcloos.com" target="_blank">cloos@jhcloos.com</a>><br>
> OpenPGP: <a href="https://jhcloos.com/0x997A9F17ED7DAEA6.asc" target="_blank">https://jhcloos.com/0x997A9F17ED7DAEA6.asc</a><br>
<br>
--<br>
Alex Balashov<br>
Principal Consultant<br>
Evariste Systems LLC<br>
Web: <a href="https://evaristesys.com" target="_blank">https://evaristesys.com</a><br>
Tel: +1-706-510-6800<br>
<br>
_______________________________________________<br>
VoiceOps mailing list<br>
<a href="mailto:VoiceOps@voiceops.org" target="_blank">VoiceOps@voiceops.org</a><br>
<a href="https://puck.nether.net/mailman/listinfo/voiceops" target="_blank">https://puck.nether.net/mailman/listinfo/voiceops</a></p>
</blockquote>
</div>
<div>
<div>
<div>
<div>
<div>
<div>
<div>
<div>
<div>
<div>
<div>
<div>
<div>
<div>
<div>
<p class="MsoNormal"><b>Pinchas S. Neiman</b><br>
Software Engineer At ESEQ Technology Corp.<br>
845.213.1229 #2</p>
</div>
<p class="MsoNormal"><img border="0" height="68" id="_x0000_i1025" src="https://ci3.googleusercontent.com/mail-sig/AIorK4z1Lx063u893FlkIV1C3aJbVPjgKnaA2Xu8q_iPdyFOnK_JX05usgghpAIwmPqB-1t-3fdaShNHoCPf7fFwa1twYZt-xjsBZheqmsCQrg" width="200"></p>
<p> </p>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
<p><br>
<br>
<br type="_moz"></p>
</body>
</html>