[cisco-voip] Force an outgoing call through a subscriber

[ROZA, Ariel]

The cluster is on version 9.1.2

De: bmeade90 at gmail.com [mailto:bmeade90 at gmail.com] En nombre de Brian Meade
Enviado el: lunes, 24 de julio de 2017 04:51 p.m.
Para: ROZA, Ariel <Ariel.ROZA at LA.LOGICALIS.COM>
CC: Ryan Huff <ryanhuff at outlook.com>; Anthony Holloway <avholloway+cisco-voip at gmail.com>; cisco-voip <cisco-voip at puck.nether.net>
Asunto: Re: [cisco-voip] Force an outgoing call through a subscriber

Ariel,

What version are you on?  There were some VMWare driver issues a long time ago that would cause issues with IPVMS resources- https://bst.cloudapps.cisco.com/bugsearch/bug/CSCtz29142<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>

Brian

On Mon, Jul 24, 2017 at 3:31 PM, ROZA, Ariel <Ariel.ROZA at la.logicalis.com<mailto:Ariel.ROZA at la.logicalis.com>> wrote:
Hi, Ryan, Anthony, et al,

  I´m sorry by the blunt nature of my first mail. I was burned out, at my customer´s.

  My problem is this:
  The users are complaining of broken audio in random calls to the PSTN

  We Isolated the issue to a single subscriber that acts as MTP for calls, sometimes.
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.
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.
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

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)


De: Ryan Huff [mailto:ryanhuff at outlook.com<mailto:ryanhuff at outlook.com>]
Enviado el: domingo, 23 de julio de 2017 07:03 p.m.
Para: Anthony Holloway <avholloway+cisco-voip at gmail.com<mailto:avholloway%2Bcisco-voip at gmail.com>>
CC: ROZA, Ariel <Ariel.ROZA at LA.LOGICALIS.COM<mailto:Ariel.ROZA at LA.LOGICALIS.COM>>; cisco-voip <cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>>
Asunto: Re: [cisco-voip] Force an outgoing call through a subscriber

Anthony,

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.

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).

Always happy to split hairs :).

Sent from my iPhone

On Jul 23, 2017, at 5:18 PM, Anthony Holloway <avholloway+cisco-voip at gmail.com<mailto:avholloway+cisco-voip at gmail.com>> wrote:
This may be splitting hair here, but two things Ryan:

1) Your first sentence reads to me like a contradiction.  Could you clarify what you're stating here?

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.

On Fri, Jul 21, 2017 at 2:53 PM Ryan Huff <ryanhuff at outlook.com<mailto:ryanhuff at outlook.com>> wrote:
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.

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).

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.

Is the problem you're troubleshooting have anything to do with one-way or no-way audio by chance?

Thanks,

Ryan

On Jul 21, 2017, at 3:37 PM, ROZA, Ariel <Ariel.ROZA at LA.LOGICALIS.COM<mailto:Ariel.ROZA at LA.LOGICALIS.COM>> wrote:
Hi, Guys.
I need to test problems with calls outgoing from an Ip phone to the PSTN  through a particular subscriber (as MTP?).
How can I force them to do that.
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.
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