RE: [AVT] RE: Dynamic selection of Codec??

"Desineni, Harikishan" <hdesinen@qualcomm.com> Thu, 29 March 2007 17:42 UTC

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Subject: RE: [AVT] RE: Dynamic selection of Codec??
Date: Thu, 29 Mar 2007 10:41:37 -0700
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From: "Desineni, Harikishan" <hdesinen@qualcomm.com>
To: "Ingemar Johansson S (LU/EAB)" <ingemar.s.johansson@ericsson.com>, avt@ietf.org
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There are few more speech/audio codecs that can be considered (from the EVRC codec family).

EVRCB:
Narrowband speech codec (8kHz sample rate)
Source controlled variable rate codec
Bit rates 0.8Kbps-8.55kbps
(aka the codec used in 3GPP2 networks with multiple capacity operating points)
Payload formats in RFC 4788
Published spec is available at http://www.3gpp2.org/Public_html/specs/C.S0014-B_v1.0_060501.pdf

EVRC-WB: 
Widedband speech codec (16kHz sample rate)
Source controlled variable rate codec
Bit rates 4 0.8Kbps-8.55Kbps,
Wideband quality even at an average bit-rate 7.5Kbps
Payload format in draft-ietf-avt-rtp-evrc-wb-00.txt
Published spec is available at http://www.3gpp2.org/Public_html/specs/C.S0014-C_v1.0_070116.pdf

-Hari

> -----Original Message-----
> From: Ingemar Johansson S (LU/EAB)
> [mailto:ingemar.s.johansson@ericsson.com]
> Sent: Tuesday, March 27, 2007 10:59 PM
> To: avt@ietf.org
> Cc: annasagaram.vamsi@aftek.com
> Subject: [AVT] RE: Dynamic selection of Codec??
> 
> Hi
> 
> As I forgot to change the subject line in my last email I need to repost
> it, sorry for the double posting
> 
> There are a few speech/audio codecs available that support dynamic changes
> in bitrate during a session (without the need for a SIP reINVITE). From
> the AMR codec family one can chose from
> 
> AMR:
>   Narrowband speech codec (8kHz sample rate).
>   Bitrates 4.75-12.2kbps
>   (aka the codec used in GSM, UMTS)
>   http://en.wikipedia.org/wiki/AMR-WB
> AMR-WB:  Widedband speech codec (16kHz sample rate).
>   Bitrates 6.60-23.85kbps
>   http://en.wikipedia.org/wiki/AMR-WB
> Payload format in RFC3267 and soon updated by draft-ietf-avt-rtp-amr-bis
> 
> AMR-WB+:
>   Speech and audio stereo/mono codec.
>   Audio bandwidth varies between 6.4kHz and 20kHz depending on
>   bitrate (5.2-48kbps), the sample rate remains the same though due to
>   internal "dynamically adjustable" resampling. Worth notice is also
>   that it is possible to even switch between stereo and mono
>   almost seamlessly during a session
>   http://en.wikipedia.org/wiki/AMR-WB%2B
> Payload format in RFC4352
> 
> C-code for the codecs mendioned above can be downloaded from
> http://www.3gpp.org/ftp/Specs/html-info/26-series.htm
> The floating point c-code are specs 26.104, 26.204 and 26.304
> Also: The bitrate switching and the associated signaling is specified in
> 3GPP TS26.114 available from the same website, this does actually adress
> the dynamic codec rate question described below.
> 
> Regards
> Ingemar
> 
> > -----Original Message-----
> > From: avt-request@ietf.org [mailto:avt-request@ietf.org]
> > Sent: den 27 mars 2007 16:41
> > To: avt@ietf.org
> > Subject: avt Digest, Vol 35, Issue 30
> >
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> > Today's Topics:
> >
> >    1. RE: Dynamic selection of Codec?? (annasagaram.vamsi@aftek.com)
> >    2. RE: Dynamic selection of Codec?? (annasagaram.vamsi@aftek.com)
> >    3. RE: Dynamic selection of Codec?? (Even, Roni)
> >
> >
> > ----------------------------------------------------------------------
> >
> > Message: 1
> > Date: Tue, 27 Mar 2007 20:05:08 +0530
> > From: <annasagaram.vamsi@aftek.com>
> > Subject: RE: [AVT] Dynamic selection of Codec??
> > To: "'Taddei, Herve'" <herve.taddei@siemens.com>,	<avt@ietf.org>
> > Message-ID: <mailman.101728.1175006465.2710.avt@ietf.org>
> > Content-Type: text/plain;	charset="iso-8859-1"
> >
> > Hi,
> >
> > Is this only limited for wideband / scalable codecs?? If
> > that's the case, then how can we it be implemented if I don't
> > have scalable codecs?
> >
> > Thanks & regards,
> > Vamsi.
> >
> > -----Original Message-----
> > From: Taddei, Herve [mailto:herve.taddei@siemens.com]
> > Sent: Tuesday, March 27, 2007 7:03 PM
> > To: annasagaram.vamsi@aftek.com; avt@ietf.org
> > Subject: AW: [AVT] Dynamic selection of Codec??
> >
> > Hi,
> >
> > You can use a scalable codec for example G.729.1 standardized
> > in the ITU-T and decide on the bitrate you want to say based
> > on the network conditions.
> > You don't need any codec re-negotiation when you change of
> > bitrate. RTP payload format is RFC 4749.
> >
> > Cheers
> >
> > Hervé
> >
> > -----Ursprüngliche Nachricht-----
> > Von: annasagaram.vamsi@aftek.com [mailto:annasagaram.vamsi@aftek.com]
> > Gesendet: Dienstag, 27. März 2007 15:14
> > An: avt@ietf.org
> > Betreff: [AVT] Dynamic selection of Codec??
> >
> > Hello,
> >    Can we select codec dynamically in the mid session when a
> > SIP call is going on?
> >
> > Suppose in 10 min. of a SIP call, Initially my bandwidth is
> > excellent so I choose for quality codec rather than bandwidth
> > consumption codec, and after few minutes my bandwidth is very
> > less, so at that time, how can I manage my quality of call?
> >
> 
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