RE: [Sipping] Use cases of multi-transcoding

"Henry Sinnreich" <henry@pulver.com> Tue, 07 March 2006 14:23 UTC

Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1FGd6Q-0002Rn-Hk; Tue, 07 Mar 2006 09:23:54 -0500
Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1FGd6O-0002Pq-5e for sipping@ietf.org; Tue, 07 Mar 2006 09:23:52 -0500
Received: from mail.pulver.com ([192.246.69.184]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1FGd6L-0004V1-DZ for sipping@ietf.org; Tue, 07 Mar 2006 09:23:52 -0500
Received: (qmail 21993 invoked by uid 510); 7 Mar 2006 09:57:56 -0500
Received: from henry@pulver.com by mail.pulver.com by uid 508 with qmail-scanner-1.22-st-qms (clamdscan: 0.72. spamassassin: 2.63. Clear:RC:1(24.1.142.176):. Processed in 1.180761 secs); 07 Mar 2006 14:57:56 -0000
X-Antivirus-MYDOMAIN-Mail-From: henry@pulver.com via mail.pulver.com
X-Antivirus-MYDOMAIN: 1.22-st-qms (Clear:RC:1(24.1.142.176):. Processed in 1.180761 secs Process 21986)
Received: from c-24-1-142-176.hsd1.tx.comcast.net (HELO 1AB764895C324D3) (henry@pulver.com@24.1.142.176) by 192.246.69.184 with SMTP; Tue, 07 Mar 2006 14:57:55 +0000
From: Henry Sinnreich <henry@pulver.com>
To: 'Kang Tae-Gyu' <tgkang@etri.re.kr>, Albrecht.Schwarz@alcatel.de
Subject: RE: [Sipping] Use cases of multi-transcoding
Date: Tue, 07 Mar 2006 08:23:43 -0600
Organization: pulver.com
MIME-Version: 1.0
X-Mailer: Microsoft Office Outlook, Build 11.0.6353
X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2900.2180
Thread-Index: AcZBuQKVV/f1qX48Siu0dEYcRw1wwAANrpWA
In-Reply-To: <8ce001c641b9$02985250$8410fe81@etri.info>
X-Antivirus-MYDOMAIN-Message-ID: <114174347583521986@mail.pulver.com>
X-Spam-Score: 0.1 (/)
X-Scan-Signature: 58f9ad36a39c86859dabf772b2ea7a96
Cc: sipping@ietf.org
X-BeenThere: sipping@ietf.org
X-Mailman-Version: 2.1.5
Precedence: list
Reply-To: henry@pulver.com
List-Id: "SIPPING Working Group \(applications of SIP\)" <sipping.ietf.org>
List-Unsubscribe: <https://www1.ietf.org/mailman/listinfo/sipping>, <mailto:sipping-request@ietf.org?subject=unsubscribe>
List-Post: <mailto:sipping@ietf.org>
List-Help: <mailto:sipping-request@ietf.org?subject=help>
List-Subscribe: <https://www1.ietf.org/mailman/listinfo/sipping>, <mailto:sipping-request@ietf.org?subject=subscribe>
Content-Type: multipart/mixed; boundary="===============1255019006=="
Errors-To: sipping-bounces@ietf.org
Message-Id: <E1FGd6Q-0002Rn-Hk@megatron.ietf.org>

Kang,

 

It seems there are several issues in this debate (in increasing order of
disruption :-) ):

 

1.	Are there too many codecs around? If yes, what is the minimum number
of codecs that should be supported? What is the recommended default codec?
(As mentioned, my pick is iLBC as the default and SPEEX for wideband).

 

2.	Can transcoding between domains be avoided or at least minimized?
(Hint: Avoided at all cost).

 

3.	Why are intermediate VoIP domains necessary at all, when e2e
services like Skype have proven to be successful and having the best audio
quality (apart from the e2e QoS debate)? Editorial: Intermediate VoIP
domains are IMHO a revenue objective, but have no technical justification on
the e2e Internet and no amount of politics can prevail in the long term over
sound technology.

Thanks, Henry


  _____  

From: Kang Tae-Gyu [mailto:tgkang@etri.re.kr] 
Sent: Tuesday, March 07, 2006 1:30 AM
To: Henry Sinnreich; Albrecht.Schwarz@alcatel.de
Cc: sipping@ietf.org
Subject: RE: [Sipping] Use cases of multi-transcoding

 

Hi, Henry

If industry has a specific "internet codec", SIP/SDP will be more simple
signaling protocol.

But, I believe that SIP/SDP tries to support any codec and any media.

I think that the making default codec is another issue. 

 

Thanks, 

Kang

--------------------------------------------------------
The numerous wireline legacy ITU-T G.7xx codecs and other countless codecs
deployed for mobile phones are a disgrace IMHO.

The industry would be better off adopting "Internet codecs":
The iLBC RFC 3951 and RFC 3952 as the default codec and SPEEX for variable
rate wideband
(ftp://ftp.rfc-editor.org/in-notes/internet-drafts/draft-ietf-avt-rtp-speex-
00.txt).

Thanks, Henry


-----Original Message-----
From: Albrecht.Schwarz@alcatel.de [mailto:Albrecht.Schwarz@alcatel.de]
Sent: Tuesday, February 21, 2006 8:29 AM
To: Kang Tae-Gyu
Cc: sipping@ietf.org
Subject: Re: [Sipping] Use cases of multi-transcoding


Hi Kang,

just a comment to the transcoding scenario between two different WB codecs
(WB1<->WB2).

Transcoding as such should be avoided because it is inherently decreasing
the audio/speech quality ("adds e.g., delay impairments and Ie, equipment
impairment factors") and an "expensive" network function.
There must be therefore an agreed justifaction, particularly for WB1<->WB2
transcoding scenarios!

Transcoding between two different NB codecs (NB1<->NB2) is often
inevitable.

The difference with WB codecs is that terminals "should support a fallback
mode to a NB codec" in my opinion. There are two possibilities:
      NB mode is part of the WB codec modes of operation (e.g., G.729EV
      with G.729A as NB mode), or
      separate NB codec (e.g., terminal concept according G.725).

The inconsistent "WB1<->WB2" E2E scenario might then lead to fallbacks to
(default) NB codecs. In case of a then "NB1<->NB2" inconsistency to NB
transcoding in the network.

Comments?

Albrecht






                      "Kang Tae-Gyu"

                      <tgkang@etri.re.         To:      <sipping@ietf.org>

                      kr>                      cc:      xupeili@huawei.com,
rohan@ekabal.com, Gonzalo.Camarillo@ericsson.com,     
                                               dean.willis@softarmor.com

                      21.02.2006 03:12         Subject: [Sipping] Use cases
of multi-transcoding                                  
                      Please respond

                      to Kang Tae-Gyu







Hello,

I would like to discuss about the use case of multi-transcoding. First of
all, I am sorry for that I could not answer about its use case when I
received the question from the floor, IETF 64 Vancouver meeting, because of
poor English. Also, thank you for the minutes(chairmen and note takers).

There are three kinds of use cases of multi-transcoding.
 -    one or two more heterogeneous networks
  -    one or two more ITSPs
  -    one or two more wideband speech codecs



There are one or two more heterogeneous networks: enterprise networks using
IP-PBX, ITSP(Internet Telephony Service Provider), IMS in 3GPP2,
PacketCable, and Wibro. They will not use a common codec. So, they need
multi transcoding such as;
 -    use case 1: A(a) - IP PBX T1(a-b) - ITSP T2(b-c, b-d) - IMS T3(a-d,
d-e) - D(d)
 -    use case 2: A(a) - IP PBX T1(a-b) - ITSP T2(b-c, b-d) - PacketCable
T4(c-d) - F(d)
 -    use case 3: E(e) - IMS T3(a-d, e-d) - ITSP T2(b-c, b-d) - Wibro
T4(a-d, d-c) - G(c)



There are one or two more ITSPs or SBC because all of internet telephony
subscribers will not subscribe only one ITSP. There are a lot of domestic
ITSPs and international ITSPs or ITXP. Internet telephony should be
supported any kind of terminal vendor even though it supports any specific
codec. It also supports multi call forwarding service. So, they need multi
transcoding such as;
 -    use case 4: C(c) - ITSP T2(c-b, b-d) - ITSP T6(b-a, b-d) - ITSP7
T7(a-d, d-e) - G(d)



There are one or two more wideband speech codecs. There has been developing
one or two more wideband speech codecs for internet telephony: AMR-WB in
3GPP, VMR-WB in 3GPP2, G.729EV in ITU-T SG 16 Question 10, a wideband codec
in ITU-T SG 16 Question 9, and AAC in ISO/IEC. In this convention, Capital
letter means a node and lowercase(a, b, c, d, and e) means a codec.

A wideband codec encodes voice using 16,000 samples per second (50 ~
7,000Hz), as opposed to the 8,000 samples per second (300 ~ 3,400Hz) by
narrowband codec. A voice quality of wideband codec is better than one of
narrowband codec due to support wider band. A tandem transcoding means to
encode narrowband codec such as G.711. So, it makes down sampling and worse
the voice quality. If two parties(calling party and called party) have two
different wideband codecs, transcoders should transcode a wideband codec to
another wideband codec without tandem transcoding. If we use a common codec
with G.711(tandem), there is no more need multi-transcoding. But, we need a
multi-transcoding for supplying a wideband speech high quality to each
sides.
 -    use case 5: wideband codec to wideband codec without tandem
transcoding



Multi transcoding signaling can be useful for internet telephony
environments supported by standard; an example VoIP network in quality of
service for next generation Voice over IP networks, MSF-TR-QoS-001.final,
2003 Feb, and Telecommunications and Internet Protocol Harmonization over
Networks (TIPHON) Release 3; End-to-End Quality of Service in TIPHON
Systems; Part 3: Signalling and Control of End-to-End Quality of Service
(QoS)-V2.1.2, 2002.

Multi trascoder can cover the use case using one or two more conference
bridge transcoding model.

We would like to discuss open these use cases or another use case. Comments
are welcome.

Thanks,
Kang


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP






_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP




 
<http://umail.etri.re.kr/External_ReadCheck.aspx?email=henry@pulver.com&name
=Henry+Sinnreich&fromemail=tgkang@etri.re.kr&messageid=%3C8ebc20e7-ea62-438c
-91a9-53bf8b088c9a@etri.re.kr%3E> 

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP