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= Dual-tone Multi-frequency Relay  =

Dual-tone Multi-frequency Relay (DTMF)&nbsp;is the mechanism where a VoIP gateway listens for in-call tones, and relay them to the peer according to the negotiated method. 

Relaying DTMF&nbsp;prevents loosing its signal integrity over VoIP compressed codecs. &nbsp;The relayed DTMF is then being regenerated transparently on the peer side. 

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There are 3 common ways of relaying DTMF: 

#[http://www.ietf.org/rfc/rfc2833.txt RFC2833] 
#SIP INFO 
#In Band<br>

== RFC2833  ==

[http://www.ietf.org/rfc/rfc2833.txt RFC2833]&nbsp;is the prefered method for relaying, where the in-call DTMF are removed from the in-band voice, and are sent to the peer over the RTP stream, as specially marked packet. 

This method also offers a robustness against packet losses by using redundancy scheme. &nbsp; 

RFC2833 is usually being handled by VoIP hardware, and requires no CPU intervention. 

== SIP INFO  ==

The SIP INFO method can be used by SIP network elements to transmit DTMF tones out-of-band as telephone-events in a reliable manner independent of the media stream. 

In the DTMF relay method the body of the SIP message consists of signaling information and uses the content-type application/dtmf-relay 

This method can be used where peer VoIP doesn't support RFC2833. This method relies on CPU power to relay the DTMF.<br> 

== In Band  ==

In-band is used when the two other methods aren't available. &nbsp;The DTMF is relay with in-band voice,&nbsp;and is more likely to work on lossless codecs, like G.711.<br> 

== Toolpack DTMF-Relay  ==

By default, Toolpack Gateway tries to negotiate RFC2833 DTMF&nbsp;relay, by announcing its telephone-event&nbsp;capability in the SIP INVITE. 

When telephone-event SDP&nbsp;negotiation fails, then SIP INFO is used. &nbsp;When RFC2833 is used, SIP INFO DTMF-Relay events are not relayed. 

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This default behavior can be overriden using a&nbsp;[[Toolpack:SIP Profiles|SIP Profile]]. &nbsp;DTMF-Relay methods can be added/removed and ordered as wanted.<br>