Cisco Cisco MeetingPlace Gateway SIM Anleitung Für Quick Setup

Enable RFC 2833 detection

There are three DTMP methods:

RFC-2833, which is negotiated and can be disabled.

KPML, which is negotiated and cannot be disabled

In-band DTMF tones, which is not negotiated but can be disabled (see"Enable 
in-band DTMF detection" setting)

RFC 2833 is a standard mechanism for transmitting keypad digits in-bandin VoIP 
media packets. It is commonly used as an adjunct to SIP signaling.Most calls will 
negotiate either RFC 2833 (in band) or KPML

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 (out of band) depending on the 

capabilities of the user endpoint. 

If both RFC-2833 and KPML are negotiated (implying that RFC-2833 was enabled), 
Cisco Unified MeetingPlace will listen for RFC-2833 and not KPML. You can force the 
use of KPML by disabling RFC 2833 if you are trying to validate KPML. Otherwise, 
disabling RFC-2833 is typically not necessary as most calls will not notice a difference. 
If you do notice a difference it may be due to Cisco Unified Communications Manager 
inserting a MTP to translate RFC-2833 to KPML. This happens if a trunk or endpoint 
does not support out-of-band signaling. Depending on the setup, MTP insertion may 
result in loss of video or, if you run out of MTP resources, call failure. 

Default: Yes

Enable  in-band  DTMF  detection

Whether to turn on the signal processing which looks for in-band acoustic DTMF

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 tones 

in the input audio media stream. Note that DTMF works well only with the G.711 
codec.

Recommendation: Enter Yes to support terminals that lack another signaling 
mechanism, including RFC 2833, KPML, or H.245. Enter No if you find that 
Cisco Unified MeetingPlace responds to voices as if they were keypad inputs (talk off).

Default: Yes

Maximum size (milliseconds)

Minimum size (milliseconds)

Maximum and minimum lengths of time, in milliseconds, that the jitter buffer holds 
voice packets. A large jitter buffer helps the system accurately reassemble the media 
stream, but it adds to perceived latency.

Jitter refers to the variation in the delay of received packets. When voice data is sent 
across the network, the packets are sent in a continuous stream with the packets spaced 
evenly apart. Due to network congestion, improper queuing, or configuration errors, the 
delay between each received packet can vary instead of remaining constant. Some 
packets may even arrive out of order or not arrive at all. 

A higher 

 helps the system adapt to poor conditions. A 

lower value may be better for interactive conversations, where an occasional dropped 
packet may be preferable to long latency. 

The 

 is the starting jitter buffer size. The closer this value 

is to the typical jitter on the network, the quicker the system adapts, but this adds 
directly to latency.

 default: 250

 default: 30