Media Server handling of DTMF tones; RFC 2833/rfc4733/rfc4734/rfc5244

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Ivelin Ivanov

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Dec 9, 2012, 6:57:39 PM12/9/12
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After last week's discussion about DTMF detection issues raised by
Restcomm Beta 2 users, there were several offline discussions on the
subject.

Here is a summary of what I learned plus information available online
related to industry best practices:

1. RTP out of band events are the recommended way to signal DTMF tones
in IP networks.
1.a. RTP inband DTMF tones are unreliable over RTP in a number of
situations, including loss of information with compression codecs,
overlap with voice bands, background noise and others.
2.b. SIP INFO is not a preferred mechanism due to timing artifacts.

2. Typically DTMF tones are converted to and from RFC 2833 RTP events
at the border with PSTN networks.
2.a. RTP DTMF out of band events remain unchanged while they traverse
IP networks.
2.b. When converting from inband to out of band, the inband signal has
to be removed. That causes sometimes a small inband leak, because it
takes time to detect the inband DTMF tone. If the same voice stream
crosses back to the PSTN network, the small leaked tone plus the full
tone merged inband from the RTP events can sometimes cause DTMF
detection errors in IVR or voice mail equipment listening for DTMF
tones.

3. SDP negotiation can only suggest behavior related to DTMF events,
but cannot be a reliable predictor.
3.a. Due to historical reasons and behavior of older equipment, even
if an end point advertises support for out of band DTMF events, that
does not mean that the far end should rely on receiving DTMF tones
only via out of band packets.
3.b. In practice all modern RTP gateways send DTMF tones via RFC 2833.

4. RFC 4733, 4734 and 5244 clarify ambiguities in the original rfc 2833.


Here is some reference material for further reading:

3CX VoIP blog » Introduction to DTMF & SIP and RFC 2833
http://www.3cx.com/blog/voip-howto/dtmf-rfc2833/

voip - "RFC 2833 RTP Event" Consecutive Events and the E "End" Bit -
Stack Overflow
http://stackoverflow.com/questions/2767665/rfc-2833-rtp-event-consecutive-events-and-the-e-end-bit

Configuring and debugging DTMF (RFC 2833) - In the VoIP net world... -
Site Home - MSDN Blogs
http://blogs.msdn.com/b/rita_z/archive/2005/10/10/479293.aspx

[Sip-implementors] RFC 2833 SIP/SDP negotiation questions
https://lists.cs.columbia.edu/pipermail/sip-implementors/2004-June/006558.html

Fusion VoIP API Developer's Manual: Transferring DTMF digits according
to RFC 2833
http://www.dialogic.com/webhelp/NaturalAccess/Release9.0/Fusion_VoIP_API_Dev_Manual/Transferring_DTMF_digits_according_to_RFC_2833.htm

DTMF issues and problems when using VoIP.
http://www.voipmechanic.com/dtmf-issues.htm



Ivelin
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