[VOIPSEC] Really RTP is end-to-end?

[Wence Van der Meersch]

Yes, there is rfc2833 (through the RTP payload) and rfc2976 (using SIP
INFO messages).
Since the original post talks about asterisk,
http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode can be
interesting.
For rfc2833 asterisk needs to be in the signal path as well since the
digits are contained in the RTP stream. The SIP INFO method often fails
to get through when your conversation passes a few hops on differently
configured systems. 

Wence Van der Meersch
Information Security Consultant
Ascure NV

e-mail 	wence.vandermeersch at ascure.com
Web    	http://www.ascure.com/
 

> -----Original Message-----
> From: Simon Horne [mailto:s.horne at packetizer.com] 
> Sent: woensdag 19 juli 2006 3:11
> To: Wence Van der Meersch
> Cc: voipsec at voipsa.org
> Subject: Re: [VOIPSEC] Really RTP is end-to-end?
> 
> 
>  From a SRTP standpoint having DTMF digits carried inband 
> (RFC 2833) is less than ideal as each hop would need to 
> decrypt and re-encrypt the media at each hop just to detect 
> the presence of DTMF digits. Is there a way to carry DTMF 
> digits out of band like other protocols can do? I mean you 
> can encrypt media right through the asterisk box in H.323
> 
> Simon
> 
> 
> At 08:24 PM 18/07/2006, you wrote:
> >This is the default asterisk behaviour, it keeps itself in 
> the RTP data 
> >path to listen for inband DTMF for feature codes (eg. for recording, 
> >transferring, ...). This also allows it to transcode between 
> different 
> >codecs.
> >
> >Regards,
> >
> >Wence Van der Meersch
> >Information Security Consultant
> >Ascure NV
> >
> >e-mail  wence.vandermeersch at ascure.com
> >Web     http://www.ascure.com/
> >
> >
> > > -----Original Message-----
> > > From: Voipsec-bounces at voipsa.org
> > > [mailto:Voipsec-bounces at voipsa.org] On Behalf Of rgigli at libero.it
> > > Sent: maandag 17 juli 2006 9:32
> > > To: voipsec
> > > Subject: [VOIPSEC] Really RTP is end-to-end?
> > >
> > > Hi,
> > > I have two x-lite softphone, A and B, and an asterisk server, C, 
> > > when A call B initiates a SIP session whit C. My problem 
> is in RTP 
> > > session because A and B don't communicate in end-to-end 
> manner but 
> > > through C. I think there is a problem in my x-lite 
> configuration, is 
> > > it right?
> > >
> > >
> > > ______________________________________________________________
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> al giorno, e 
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> > >
> > >
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