[SIPForum-discussion] What is the function of telephone-event?
Jason Nesheim
jnesheim at telepacific.com
Thu Mar 6 18:27:12 UTC 2008
If you are not using a media proxy of some sort, ie a SBC, the RTP flows
directly between endpoints. In that case the proxy never receives the
RFC2833 telephone events so it is not possible for it to convert those
events to a SIP INFO event. At that point it would be a requirement for
both of the SIP endpoints to support RFC2833.
On 3/6/08 9:46 AM, "Victor Pascual Ávila" <victor.pascual.avila at gmail.com>
wrote:
> Folks,
is there any interoperability between DTMF modes? If I'm using
RFC2833,
> is there other possibility to convert it to SIP Info Method
not using a
> media-proxy?
Cheers,
Victor Pascual
On Thu, Mar 6, 2008 at 5:18 PM, Andrea
> Puddu <androjoker at hotmail.com> wrote:
>
> It's clear now. Thanks
> everybody.
>
> Andrea
>
>
>
> ________________________________
> From:
> mh at markholloway.com
> To: androjoker at hotmail.com; steve.langstaff at citel.com;
>
> discussion at sipforum.org
>
> Subject: RE: [SIPForum-discussion] What is the
> function of telephone-event?
> Date: Thu, 6 Mar 2008 08:08:07
> -0700
>
>
>
>
>
>
> G729 does not support inband DTMF because the codec
> compresses the audio.
> You must use out of band. Therefore, telephone-event
> is used to tell the
> other endpoint how it should receive DTMF tones. If you
> are using G711 you
> should be able to support inband. If you are using a
> compressed codec you
> should consider using out of band. This can be RFC
> 2833 or SIP Info Method.
> Regardless, not supporting telephone-event reflects
> a poor SIP
> implementation in my opinion.
>
>
>
>
>
>
>
> From:
> discussion-bounces at sipforum.org
> [mailto:discussion-bounces at sipforum.org] On
> Behalf Of Andrea Puddu
> Sent: Thursday, March 06, 2008 7:09 AM
> To: Steve
> Langstaff; discussion at sipforum.org
> Subject: Re: [SIPForum-discussion] What
> is the function of telephone-event?
>
>
>
> Thanks Steve.
>
> Why did you
> talk about low bit-rate codec? I understood the matter about
> dtfm out of
> band, but it is not clear to me why it should impact low bit
> rate codec
> (e.g. g729).
>
> Thanks,
>
> Andrea
>
>
>
>
>
> ________________________________
>
>
> Subject: RE: [SIPForum-discussion] What
> is the function of telephone-event?
> Date: Thu, 6 Mar 2008 05:01:34 -0800
>
> From: steve.langstaff at citel.com
> To: androjoker at hotmail.com;
> discussion at sipforum.org
>
> "telephone-event" is a method for passing e.g.
> DTMF signalling in an RTP
> stream.
>
>
>
> The "101" (or any payload type in
> the range 96-127) refers to a dynamic
> payload type.
>
>
>
> So a stream that
> negotiates to use "101 telephone-event" may e.g. pass
> keypresses
>
> using
> RTP payload type 101 rather than "in-band".
>
>
>
> The risks you face in
> removing telephone event support are that if you are
> using a
>
> low bitrate
> codec such as G.729, or you are using equipment that does not
> expect
>
> to
> see DTMF "in-band" then users may lose the ability to control e.g. IVR
>
> equipment.
>
>
>
> ________________________________
>
>
> From:
> discussion-bounces at sipforum.org
> [mailto:discussion-bounces at sipforum.org] On
> Behalf Of Andrea Puddu
> Sent: 06 March 2008 12:15
> To:
> discussion at sipforum.org
> Subject: [SIPForum-discussion] What is the function
> of telephone-event?
>
> Hello guys,
>
> as the network here does not
> completely support "telephone event" in SDP
> they asked me to set up phones
> to not include "telephone event".
> I have looked at some RFCs .... but I
> couldn't understand what the
> telephone events are useful to.
>
> Do you see
> any risk to remove the telephone event, in particular the 101
> telephone
> event?
>
> thanks 1000.
>
> Andrea
>
>
>
> ________________________________
>
>
> MSN Video Interviste, concerti, news e
> videoclip! Solo su MSN Video!
>
>
> ________________________________
>
>
>
> MSN Video Interviste, concerti, news e videoclip! Solo su MSN Video!
>
>
>
> ________________________________
> MSN Video Interviste, concerti, news e
> videoclip! Solo su MSN Video!
>
> _______________________________________________
> This is the SIP Forum
> discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options,
> please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the
> list at discussion at sipforum.org
>
>
--
Victor Pascual Ávila
Research
> Engineer
Tel. +34 93 542 2906
Fax. +34 93 542 2517
Research Group on Network
> Technologies and Strategies (NeTS)
Universitat Pompeu Fabra (UPF)
Pg. de
> Circumval·lació, 8
Office 358
08003 Barcelona
> (Spain)
http://nets.upf.edu/
_______________________________________________
> This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your
> delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
Post to the list at
> discussion at sipforum.org
--
Jason Nesheim, Network Engineer
Telepacific Communications
Office: 702-405-3571
Mobile: 702-885-0815
More information about the discussion
mailing list