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

[Brian Meade]

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> 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
>   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
> Building 22 Room 131
> Work 5033995294 <(503)%20399-5294>
> Mobile 9712182110 <(971)%20218-2110>
> SIP 5035406686 <(503)%20540-6686>
>
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>
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