[cisco-voip] MTP or not MTP on sip trunk

[Ryan Huff]

Here here!

MTP is basically creating a known rendezvous point that both call legs can reach and there is a guard at the gate that says, “pay no mind to the man behind the curtain, it’s just magic”. traces and debugs would show you what’s going on but you got to go through a ton more effort to get an RCA.

Sent from my iPhone

On Feb 22, 2018, at 14:02, Brian Meade <bmeade90 at vt.edu<mailto:bmeade90 at vt.edu>> wrote:

We'll really need a "debug ccsip messages" for an entire good/bad call to see what part the carrier is getting hung up on.  MTP Required would probably be a good fix but it's a bit bad practice to rely on MTPs where not necessary.  MTPs definitely make troubleshooting more difficult.

On Thu, Feb 22, 2018 at 1:22 PM, Jonatan Quezada <jonatan.quezada at chemeketa.edu<mailto:jonatan.quezada at chemeketa.edu>> wrote:
and if so, do I need to modify this portiopn of my cube configuration?, This guy seems to describe what I am seeing on our main campus

https://supportforums.cisco.com/t5/video-over-ip/sip-trunk-call-hold-fails-no-audio-no-resume-no-sdp-from-cube/td-p/2214595

!!!!!!!!!!!!!!!!!!!

voice service voip
 address-hiding
 dtmf-interworking rtp-nte
 mode border-element license capacity 500
 media bulk-stats
 allow-connections sip to sip
 no supplementary-service sip moved-temporarily
 redirect ip2ip
 fax protocol t38 nse version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
 sip
  rel1xx supported "rel100"
  session refresh
  asserted-id pai
  privacy pstn
  localhost dns:voip.centurylink.com<http://voip.centurylink.com>
  no update-callerid
  early-offer forced
  midcall-signaling passthru
......
begin my notes:

here is where i think i should change to read like so : "midcall-signalling black" instead of passthru

4431-voice-gw(conf-serv-sip)#mid
4431-voice-gw(conf-serv-sip)#midcall-signaling ?
  block           Block all SIP messages in midcall
  passthru        Passthrough SIP messages from one IP leg to another IP leg
  preserve-codec  preserve initial negotiated codec i.e. midcall codec change
                  denial

.....
  privacy-policy passthru
  pass-thru subscribe-notify-events all
  pass-thru content sdp
  sip-profiles 100
  no call service stop
!
here is my call flow

ISP -----Their SIPtrunk----MyCube----CUCM10.5----VoiceVlan-----7975, and 8861's

the phones here and there are experiencing, one way audio, and transfer drops, not all of the time though. what do we suggest?
why would you want to block SDP updates mid call, how will CUCM know how to changes call characteristics?

please help


Johnny Q
Voice Technology Analyst - TelNet
Chemeketa Community College
Johnny.Q at chemeketa.edu<mailto:Johnny.Q at chemeketa.edu>
Building 22 Room 131
Work 5033995294<tel:(503)%20399-5294>
Mobile 9712182110<tel:(971)%20218-2110>
SIP 5035406686<tel:(503)%20540-6686>

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