[cisco-voip] SIP peers to UCM and DTMF

[Pawlowski, Adam]

Is rtp-nte the recommended configuration, if you're working in a mixed Cisco SCCP environment? Especially given that things seem to be moving forward to SIP for the latest 7900 endpoints, you'd think it would be best, but we still have some older hardware hanging out too.

Adam

> -----Original Message-----
> From: Brian Meade (brmeade) [mailto:brmeade at cisco.com]
> Sent: Wednesday, February 05, 2014 1:08 PM
> To: Eric Pedersen; Pawlowski, Adam; cisco-voip at puck.nether.net
> Subject: RE: SIP peers to UCM and DTMF
> 
> Eric,
> 
> It depends on what your endpoints support.  Most endpoints support
> RFC2833 so no need to pull in an MTP for those.  Some endpoints only
> support out-of-band methods such as H.323 gateways where an MTP would
> need to be pulled in for that case.
> 
> Brian
> 
> -----Original Message-----
> From: Eric Pedersen [mailto:PedersenE at bennettjones.com]
> Sent: Wednesday, February 05, 2014 1:01 PM
> To: Brian Meade (brmeade); Pawlowski, Adam; cisco-voip at puck.nether.net
> Subject: RE: SIP peers to UCM and DTMF
> 
> We use sip-kpml. Doesn't NTE always invoke an MTP because it's inband?
> 
> -----Original Message-----
> From: cisco-voip [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> Brian Meade (brmeade)
> Sent: 04 February 2014 2:47 PM
> To: Pawlowski, Adam; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] SIP peers to UCM and DTMF
> 
> Adam,
> 
> Some of the phone models don't play the RFC 2833 inband DTMF tones
> audibly so that may be what you're seeing.  You'll probably need a packet
> capture to ensure the RTP events are being sent to the phone.
> 
> You shouldn't need to use MTP Required as an MTP will be dynamically
> pulled in for a DTMF mismatch.
> 
> Brian
> 
> -----Original Message-----
> From: cisco-voip [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> Pawlowski, Adam
> Sent: Tuesday, February 04, 2014 4:29 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] SIP peers to UCM and DTMF
> 
> Sorry if this one has been asked recently, but, what's the recommended
> configuration of the peers/trunk for SIP between an ISR gateway and a UCM
> (v9.1)?
> 
> I have it set as rtp-nte which appears to work from the desk set -> pstn with
> no trouble, and PSTN inbound to Unity Connection or UCCX. I can see the
> messages generated in debug when I hit DTMF from the PSTN towards the
> desk set but no tone/beep is generated there. As well, I read that the older
> phones do not support rtp-nte, so I would need to have MTP required
> checked on the trunk for all calls?
> 
> Just trying to make sure I have this all straight.
> 
> Thanks,
> 
> Adam P
> SUNYAB
> 
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