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

[Kent Roberts]

Hey Jon had the exact same issue with the same carrier.  
Because they use mpls the qos policy was not correct on the carrier side so we exceeded it at times    Stupid if you ask me when the line is dedicated for voice

Stay away from mtp if you can. The gateway config should be set to the best effort mode so cucm only has to use them if it really has to.   If your just doing 711/729 and nte for dtmf you will just be adding more headaches to the troubleshooting process

If you can keep the mtp off the cube


Kent

> On Feb 22, 2018, at 11:22, 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
> Mobile 9712182110
> SIP 5035406686
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