DTMF-Relay - Revision history

Sylvain Fontaine at 20:21, 17 March 2010

2010-03-17T20:21:38Z

Sylvain Fontaine at 20:21, 17 March 2010

2010-03-17T20:21:19Z

Sylvain Fontaine at 13:59, 17 March 2010

2010-03-17T13:59:30Z

Sylvain Fontaine at 13:09, 17 March 2010

2010-03-17T13:09:20Z

Sylvain Fontaine at 13:08, 17 March 2010

2010-03-17T13:08:26Z

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.

@@ Line 17: / Line 17: @@
 [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  ==
@@ 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
 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.
+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>
-−
 <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.
-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>
+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>

@@ Line 17: / Line 17: @@
 [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.
+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.
@@ Line 23: / Line 23: @@
 == 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-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.
-−
-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. &nbsp;The DTMF is relay with in-band voice,&nbsp;and is more likely to work on lossless codecs, like G.711.
+== Toolpack DTMF-Relay  ==
-&nbsp;
+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>

@@ Line 9: / Line 9: @@
 There are 3 common ways of relaying DTMF:
 #[http://www.ietf.org/rfc/rfc2833.txt RFC2833]
 #SIP INFO
-#In Band
+#In Band<br>
-−
 <br>
 == 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]&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.
 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.
@@ 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
-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.

← Older revision		Revision as of 13:09, 17 March 2010
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	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.
		+
		+
		+
		+	****
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		+	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