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<p class="MsoNormal"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D;mso-fareast-language:EN-US">Yes, I got several captures. Some that fail and some that do not.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D;mso-fareast-language:EN-US">At the moment the wireshark analisys shows failing samples show some “jitter drops” but not packet loss.<br>
But all the failing ones go through the subscriber.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D;mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D;mso-fareast-language:EN-US">I asked my customer to take a capture sample of a broken audio call from the subscriber and from the gateway, to see if
the audio breakdown is created right on the subscriber or further down on the network.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D;mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><b><span lang="ES" style="font-size:11.0pt;font-family:"Calibri",sans-serif">De:</span></b><span lang="ES" style="font-size:11.0pt;font-family:"Calibri",sans-serif"> bmeade90@gmail.com [mailto:bmeade90@gmail.com]
<b>En nombre de </b>Brian Meade<br>
<b>Enviado el:</b> lunes, 24 de julio de 2017 06:44 p.m.<br>
<b>Para:</b> ROZA, Ariel <Ariel.ROZA@LA.LOGICALIS.COM><br>
<b>CC:</b> Ryan Huff <ryanhuff@outlook.com>; Anthony Holloway <avholloway+cisco-voip@gmail.com>; cisco-voip <cisco-voip@puck.nether.net><br>
<b>Asunto:</b> Re: [cisco-voip] Force an outgoing call through a subscriber<o:p></o:p></span></p>
<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">Probably not that bug then. Are you able to get a capture of a call using this as an MTP? Is it intermittent or always happens on this sub? <o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">I'd check the delta on the Wireshark RTP Analysis to make sure it's not jumping around between 16ms and 24ms and stays steady at 20ms.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">On Mon, Jul 24, 2017 at 5:29 PM, ROZA, Ariel <<a href="mailto:Ariel.ROZA@la.logicalis.com" target="_blank">Ariel.ROZA@la.logicalis.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"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D">The cluster is on version 9.1.2</span><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D"> </span><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><b><span lang="ES" style="font-size:11.0pt;font-family:"Calibri",sans-serif">De:</span></b><span lang="ES" style="font-size:11.0pt;font-family:"Calibri",sans-serif">
<a href="mailto:bmeade90@gmail.com" target="_blank">bmeade90@gmail.com</a> [mailto:<a href="mailto:bmeade90@gmail.com" target="_blank">bmeade90@gmail.com</a>]
<b>En nombre de </b>Brian Meade<br>
<b>Enviado el:</b> lunes, 24 de julio de 2017 04:51 p.m.<br>
<b>Para:</b> ROZA, Ariel <<a href="mailto:Ariel.ROZA@LA.LOGICALIS.COM" target="_blank">Ariel.ROZA@LA.LOGICALIS.COM</a>><br>
<b>CC:</b> Ryan Huff <<a href="mailto:ryanhuff@outlook.com" target="_blank">ryanhuff@outlook.com</a>>; Anthony Holloway <<a href="mailto:avholloway%2Bcisco-voip@gmail.com" target="_blank">avholloway+cisco-voip@gmail.com</a>>; cisco-voip <<a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a>></span><o:p></o:p></p>
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<p class="MsoNormal"><br>
<b>Asunto:</b> Re: [cisco-voip] Force an outgoing call through a subscriber<o:p></o:p></p>
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<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" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Ariel,<o:p></o:p></p>
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<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" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">What version are you on? There were some VMWare driver issues a long time ago that would cause issues with IPVMS resources- <a href="https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fbst.cloudapps.cisco.com%2Fbugsearch%2Fbug%2FCSCtz29142&data=02%7C01%7CAriel.ROZA%40la.logicalis.com%7Ccfbdb5b25a9b4fe2583708d4d2cd4b55%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636365226557540072&sdata=zUx6tJy%2BAf34fNymMdO%2BbqrJkLkJCOs3rk3%2FSkjVCE8%3D&reserved=0" target="_blank">https://bst.cloudapps.cisco.com/bugsearch/bug/CSCtz29142</a><o:p></o:p></p>
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<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" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Brian<o:p></o:p></p>
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<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" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">On Mon, Jul 24, 2017 at 3:31 PM, ROZA, Ariel <<a href="mailto:Ariel.ROZA@la.logicalis.com" target="_blank">Ariel.ROZA@la.logicalis.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"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D">Hi, Ryan, Anthony, et al,</span><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D"> </span><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D"> I´m sorry by the blunt nature of my first mail. I was burned out, at my customer´s.</span><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D"> </span><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D"> My problem is this:</span><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D"> The users are complaining of broken audio in random calls to the PSTN
</span><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D"> </span><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D"> We Isolated the issue to a single subscriber that acts as MTP for calls, sometimes.</span><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D">Wireshark captures show us that whenever the sub acts as an intermediary, audio gets jitter
(but no drops) This is on the upstream to the PSTN. The downstream is always fine.</span><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D">We suspect of a problem in the underlying nework, as the Publisher and the Subs are wired
differently and only the Subscriber show these problems.</span><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D">The Publisher is connected to a CAT4500 switch and the Sub is connected to a NEXUS 5000
that in turn is connected to the Cat4500</span><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D"> </span><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D">I am already using MRGL/MRGs to force MTP, but sometimes the phones send the RTP directly
to the gateway, bypassing the MTP. (calling always the same number)</span><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D"> </span><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D"> </span><o:p></o:p></p>
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<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><b><span lang="ES" style="font-size:11.0pt;font-family:"Calibri",sans-serif">De:</span></b><span lang="ES" style="font-size:11.0pt;font-family:"Calibri",sans-serif"> Ryan Huff [mailto:<a href="mailto:ryanhuff@outlook.com" target="_blank">ryanhuff@outlook.com</a>]
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<b>Enviado el:</b> domingo, 23 de julio de 2017 07:03 p.m.<br>
<b>Para:</b> Anthony Holloway <<a href="mailto:avholloway%2Bcisco-voip@gmail.com" target="_blank">avholloway+cisco-voip@gmail.com</a>><br>
<b>CC:</b> ROZA, Ariel <<a href="mailto:Ariel.ROZA@LA.LOGICALIS.COM" target="_blank">Ariel.ROZA@LA.LOGICALIS.COM</a>>; cisco-voip <<a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a>><br>
<b>Asunto:</b> Re: [cisco-voip] Force an outgoing call through a subscriber</span><o:p></o:p></p>
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<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" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Anthony,<o:p></o:p></p>
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<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" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Yes, it is splitting hairs IMO :). I'm specifically talking about a typical scenario where RTP is sent from/to the phone for a connected, like-codec call that is not forcing media
termination with the next device in the call leg. <o:p></o:p></p>
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<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" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">As you and I have mentioned, there are a handful of ways to use different features and services of a CUCM server to terminate and join media streams (MTP, CFB ... etc) ... and I
considers these as ancillary service and component capabilities to the server and not a core function of the server itself (hence, media not flowing through the server, although it maybe interacting with one or more of the aforementioned media services or
components).<o:p></o:p></p>
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<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" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Always happy to split hairs :).<br>
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Sent from my iPhone<o:p></o:p></p>
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<p class="MsoNormal" style="mso-margin-top-alt:auto;margin-bottom:12.0pt"><br>
On Jul 23, 2017, at 5:18 PM, Anthony Holloway <<a href="mailto:avholloway+cisco-voip@gmail.com" target="_blank">avholloway+cisco-voip@gmail.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">This may be splitting hair here, but two things Ryan:
<o:p></o:p></p>
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<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" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">1) Your first sentence reads to me like a contradiction. Could you clarify what you're stating here?<o:p></o:p></p>
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<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" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">2) Media can flow through, and even terminate on CUCM, since things like Conference Bridges, MTPs and now the new IVR media resource, are all doing that.<o:p></o:p></p>
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<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" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">On Fri, Jul 21, 2017 at 2:53 PM Ryan Huff <<a href="mailto:ryanhuff@outlook.com" target="_blank">ryanhuff@outlook.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">So the actual media (RTP) will never flow through a CUCM server; it may however, terminate a connected media stream on a software based MTP application that CUCM runs as a service
(IP Voice Media Streaming Application). Signaling (SIP) on the other hand, will always traverse a CUCM server.<o:p></o:p></p>
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<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" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">If you see a CUCM IP address in the Audio field of the SDP, then it's likely terminating on a CUCM based MTP resource (most often, due to some differences in DTMF negotiations or
because the egress path in CUCM is required to use MTP).<o:p></o:p></p>
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<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" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">If you are trying to test a call using a CUCM MTP resource on a particular cluster node; the simplest way would be to create a new MRG/MRGL that only specifies MTP resources from
the desired cluster node and then advertise that MRGL to the phone and/or egress path to the pstn for the phone and then "require" MTP termination from the phone or egress path.<br>
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Is the problem you're troubleshooting have anything to do with one-way or no-way audio by chance?<o:p></o:p></p>
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<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><br>
Thanks,<o:p></o:p></p>
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<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" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Ryan<o:p></o:p></p>
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On Jul 21, 2017, at 3:37 PM, ROZA, Ariel <<a href="mailto:Ariel.ROZA@LA.LOGICALIS.COM" target="_blank">Ariel.ROZA@LA.LOGICALIS.COM</a>> wrote:<o:p></o:p></p>
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<span style="font-size:11.0pt;font-family:"Arial",sans-serif;color:black">Hi, Guys.
</span><o:p></o:p></p>
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<span style="font-size:11.0pt;font-family:"Arial",sans-serif;color:black">I need to test problems with calls outgoing from an Ip phone to the PSTN through a particular subscriber (as MTP?).
</span><o:p></o:p></p>
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<span style="font-size:11.0pt;font-family:"Arial",sans-serif;color:black">How can I force them to do that.
</span><o:p></o:p></p>
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<span style="font-size:11.0pt;font-family:"Arial",sans-serif;color:black">Packet captures show me that, at times, calls go from my phone to the h323 gateway and sometimes they go from my phone to the Sub and then to the gew.</span><o:p></o:p></p>
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<span style="font-size:11.0pt;font-family:"Arial",sans-serif;color:black">Obtener
<a href="https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Faka.ms%2Fghei36&data=02%7C01%7CAriel.ROZA%40la.logicalis.com%7C1e58758c6884429f5d0d08d4d2169e17%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636364441966497710&sdata=0AscsFwizg82xmGqihl67oG7G1Czd3U8w5dCXgStsqM%3D&reserved=0" target="_blank">
Outlook para Android</a></span><o:p></o:p></p>
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<span style="font-size:11.0pt;font-family:"Arial",sans-serif;color:black"> </span><o:p></o:p></p>
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<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"><o:p> </o:p></p>
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