DTMF-Relay - Revision history

Luc Morissette at 15:32, 19 October 2023

2023-10-19T15:32:54Z

← Older revision		Revision as of 11:32, 19 October 2023
Line 2:		Line 2:
	= Dual-tone Multi-frequency Relay =		= Dual-tone Multi-frequency Relay =

	Dual-tone Multi-frequency Relay (DTMF)~~ ~~is the mechanism where a VoIP gateway listens for in-call tones, and relay them to the peer according to the negotiated method.		Dual-tone Multi-frequency Relay (DTMF) 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~~ ~~prevents loosing its signal integrity over VoIP compressed codecs. ~~ ~~The relayed DTMF is then being regenerated transparently on the peer side.		Relaying DTMF prevents loosing its signal integrity over VoIP compressed codecs. The relayed DTMF is then being regenerated transparently on the peer side.

	<br>		<br>
Line 12:		Line 12:
	#[http://www.ietf.org/rfc/rfc2833.txt RFC2833]		#[http://www.ietf.org/rfc/rfc2833.txt RFC2833]
	#SIP INFO		#SIP INFO
	#In ~~Band~~<br>		#In band<br>

	== RFC2833 ==		== RFC2833 ==

	[http://www.ietf.org/rfc/rfc2833.txt RFC2833]~~ ~~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.		[http://www.ietf.org/rfc/rfc2833.txt RFC2833] 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. ~~ ~~		This method also offers a robustness against packet losses by using redundancy scheme.

	RFC2833 is usually being handled by VoIP hardware, and requires no CPU intervention.		RFC2833 is usually being handled by VoIP hardware, and requires no CPU intervention.
Line 32:		Line 32:
	== In Band ==		== In Band ==

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

	== DTMF-Relay processing ==		== DTMF-Relay processing ==
Line 41:		Line 41:

	<br>		<br>

			== SIP Profiles ==

	This default behavior can be overridden using [[Toolpack:SIP Profiles\|SIP Profile]]. <br>		This default behavior can be overridden using [[Toolpack:SIP Profiles\|SIP Profile]]. <br>

Luc Morissette: /* Toolpack DTMF-Relay */

2023-10-19T15:28:53Z

Toolpack DTMF-Relay

← Older revision		Revision as of 11:28, 19 October 2023
Line 34:		Line 34:
	In-band is used when the two other methods aren't available.  The DTMF is relay with in-band voice, and is more likely to work on lossless codecs, like G.711.<br>		In-band is used when the two other methods aren't available.  The DTMF is relay with in-band voice, and is more likely to work on lossless codecs, like G.711.<br>

	== ~~Toolpack~~ DTMF-Relay ==		== DTMF-Relay processing ==

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

	When telephone-event SDP~~ ~~negotiation fails, then SIP INFO is used. ~~ ~~When RFC2833 is used, SIP INFO DTMF-Relay events are not relayed.		When telephone-event SDP negotiation fails, then SIP INFO is used. When RFC2833 is used, SIP INFO DTMF-Relay events are not relayed. Otherwise, in band DTMF is used.

	<br>		<br>

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

			It is possible to change the default mode of operation from one network to another by assigning a different profile to each network (NAP).<br>
			For example, if one network has in band DTMF and the other has RFC2833, it will capture the in band DTMF and create RTP events on the other leg and vice-versa. <br> On ProSBC this requires a transcoding unit. <br>

Allyntree at 13:50, 14 September 2022

2022-09-14T13:50:09Z

← Older revision		Revision as of 09:50, 14 September 2022
Line 1:		Line 1:
			{{DISPLAYTITLE:DTMF Relay}}
	= Dual-tone Multi-frequency Relay =		= Dual-tone Multi-frequency Relay =

Sylvain Fontaine at 20:21, 17 March 2010

2010-03-17T20:21:38Z

← Older revision		Revision as of 16:21, 17 March 2010
Line 17:		Line 17:
	[http://www.ietf.org/rfc/rfc2833.txt RFC2833] 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.		[http://www.ietf.org/rfc/rfc2833.txt RFC2833] 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.		This method also offers a robustness against packet losses by using redundancy scheme.

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

	== SIP INFO ==		== SIP INFO ==
Line 27:		Line 27:
	In the DTMF relay method the body of the SIP message consists of signaling information and uses the content-type application/dtmf-relay		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>		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 ==

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

	<br>

	== Toolpack DTMF-Relay ==		== Toolpack DTMF-Relay ==

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

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

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

			<br>

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

Sylvain Fontaine at 20:21, 17 March 2010

2010-03-17T20:21:19Z

← Older revision		Revision as of 16:21, 17 March 2010
Line 17:		Line 17:
	[http://www.ietf.org/rfc/rfc2833.txt RFC2833] 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.		[http://www.ietf.org/rfc/rfc2833.txt RFC2833] 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.		This method also offers a robustness against packet losses by using redundancy scheme.

	RFC2833 is usually being handled by VoIP hardware, and requires no CPU intervention.		RFC2833 is usually being handled by VoIP hardware, and requires no CPU intervention.
Line 23:		Line 23:
	== SIP INFO ==		== 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 ==

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

	~~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~~.

	~~== In Band ==~~		<br>

	~~In-band is used when the two other methods aren't available.  The~~ DTMF ~~is relay with in~~-~~band voice, and is more likely to work on lossless codecs, like G.711.~~		== Toolpack DTMF-Relay ==

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

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

	~~== Toolpack DTMF-Relay ==~~

	~~Toolpack gateway relay scheme is:~~

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

Sylvain Fontaine at 13:59, 17 March 2010

2010-03-17T13:59:30Z

← Older revision		Revision as of 09:59, 17 March 2010
Line 9:		Line 9:
	There are 3 common ways of relaying DTMF:		There are 3 common ways of relaying DTMF:

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

	<br>

	== RFC2833 ==		== RFC2833 ==

	~~The prefered method for relaying is~~ [http://www.ietf.org/rfc/rfc2833.txt RFC2833], 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.		[http://www.ietf.org/rfc/rfc2833.txt RFC2833] 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.		This method also offers a robustness against packet losses by using redundancy scheme.

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

	== SIP INFO ==		== SIP INFO ==

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



	~~****~~

	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.		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.
Line 35:		Line 31:
	In the DTMF relay method the body of the SIP message consists of signaling information and uses the content-type application/dtmf-relay		In the DTMF relay method the body of the SIP message consists of signaling information and uses the content-type application/dtmf-relay

	~~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.

			== In Band ==

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



			== Toolpack DTMF-Relay ==

			Toolpack gateway relay scheme is:

Sylvain Fontaine at 13:09, 17 March 2010

2010-03-17T13:09:20Z

← Older revision		Revision as of 09:09, 17 March 2010
Line 26:		Line 26:

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



			****

			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

	Content-Type: application/dtmf-relay		Content-Type: application/dtmf-relay

Sylvain Fontaine at 13:08, 17 March 2010

2010-03-17T13:08:26Z

← Older revision		Revision as of 09:08, 17 March 2010
Line 1:		Line 1:
	= Dual-tone Multi-frequency Relay =		= Dual-tone Multi-frequency Relay =

	Dual-tone Multi-frequency Relay (DTMF) is the mechanism where a VoIP gateway listens for in-call tones, and relay them to the peer according to the negotiated method.		Dual-tone Multi-frequency Relay (DTMF) 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 prevents loosing its signal integrity over VoIP compressed codecs.  The relayed DTMF is then being regenerated transparently on the peer side.		Relaying DTMF prevents loosing its signal integrity over VoIP compressed codecs.  The relayed DTMF is then being regenerated transparently on the peer side.

			<br>

			There are 3 common ways of relaying DTMF:

	~~There are 3 common ways of relaying DTMF~~:		#[http://www.ietf.org/rfc/rfc2833.txt RFC2833]
			#SIP INFO
	#RFC2833
	#SIP INFO
	#In Band		#In Band

			<br>

			== RFC2833 ==

	== RFC2833 ==		The prefered method for relaying is [http://www.ietf.org/rfc/rfc2833.txt RFC2833], 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.

	~~The prefered~~ method ~~for relaying is RFC2833, where the in-call DTMF are removed from the in-band voice, and are sent to the peer over the RTP stream, as~~ a ~~specialized DATA~~ packet.		This method also offers a robustness against packet losses by using redundancy scheme.

	~~This method also offers a robustness against packet losses~~ by ~~using redundancy scheme~~.		RFC2833 is usually being handled by VoIP hardware, and requires no CPU intervention.

	~~RFC2833 is usually being handled by VoIP hardware, and requires no CPU intervention.~~		== SIP INFO ==

	~~== SIP INFO ==~~		This method can be used where the peer VoIP doesn't support RFC2833.  This method relies on CPU power to relay the DTMF.

	~~This method can be used where the peer VoIP doesn't support RFC2833.  This method relies on CPU power to~~ relay ~~the DTMF.~~		Content-Type: application/dtmf-relay

Sylvain Fontaine: Created page with '= Dual-tone Multi-frequency Relay = Dual-tone Multi-frequency Relay (DTMF) is the mechanism where a VoIP gateway listens for in-call tones, and relay them to the peer accor…'

2010-03-17T13:05:44Z

Created page with '= Dual-tone Multi-frequency Relay = Dual-tone Multi-frequency Relay (DTMF) is the mechanism where a VoIP gateway listens for in-call tones, and relay them to the peer accor…'

New page

= Dual-tone Multi-frequency Relay =

Dual-tone Multi-frequency Relay (DTMF) 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 prevents loosing its signal integrity over VoIP compressed codecs.  The relayed DTMF is then being regenerated transparently on the peer side.

There are 3 common ways of relaying DTMF:

#RFC2833
#SIP INFO
#In Band

== RFC2833 ==

The prefered method for relaying is RFC2833, where the in-call DTMF are removed from the in-band voice, and are sent to the peer over the RTP stream, as a specialized DATA packet.

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

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

== SIP INFO ==

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