Subject: [AVT] default packetization interval for pcmu, pcma and iLBC

From: "Murat Artun" <[email-masked]>
Date: Thu, 5 Apr 2007 16:42:44 +0300

Hello,

I want to ask this for confirmation purposes.

As I read from RFC 3551 pg 10 default packetization interval of 20 ms.
So I understand that for PCMU and PCMA  frames should defaulted to be
20 ms (160 samples). Of course this is valid for other codec types
specified to be static in RF 3551.

However, as it was discussed in the below thread in avt default
packetization interval is 30 ms for iLBC:

http://www1.ietf.org/mail-archive/web/avt/current/msg05212.html

So, can we say that for an VoIP application to be standard it should
follow these packetization interval specifications?

-- 
M u r at  A r t u n, MSc.
   Software Engineer

_______________________________________________
Audio/Video Transport Working Group
[email-masked]
https://www1.ietf.org/mailman/listinfo/avt

---

From: Randell Jesup <[email-masked]>
Date: Thu, 05 Apr 2007 10:16:04 -0400

>specified to be static in RF 3551.

I think you've over-stated what RFC 3551 says:

   The following recommendations are default operating parameters.
   Applications SHOULD be prepared to handle other values.  The ranges
   given are meant to give guidance to application writers, allowing a
   set of applications conforming to these guidelines to interoperate
   without additional negotiation.  These guidelines are not intended to
   restrict operating parameters for applications that can negotiate a
   set of interoperable parameters, e.g., through a conference control
   protocol.

   For packetized audio, the default packetization interval SHOULD have
   a duration of 20 ms or one frame, whichever is longer, unless
   otherwise noted in Table 1 (column "ms/packet").  The packetization
   interval determines the minimum end-to-end delay; longer packets
   introduce less header overhead but higher delay and make packet loss
   more noticeable.  For non-interactive applications such as lectures
   or for links with severe bandwidth constraints, a higher
   packetization delay MAY be used.  A receiver SHOULD accept packets
   representing between 0 and 200 ms of audio data.  (For framed audio
   encodings, a receiver SHOULD accept packets with a number of frames
   equal to 200 ms divided by the frame duration, rounded up.)  This
   restriction allows reasonable buffer sizing for the receiver.

>http://www1.ietf.org/mail-archive/web/avt/current/msg05212.html

Sure, because the RTP packetization RFC for iLBC says that unless you
specify (and agree) otherwise via a=fmtp, the frame interval for iLBC is 30
(note: NOT the packetization interval - you can have (almost) any integer
number of frames in a packet - see above.)  Also note that since iLBC can
have (and defaults to) a frame length of 30ms, it falls into the "20ms or
one frame, whichever is longer" case.

>follow these packetization interval specifications?

See above.  Those are recommendations, and at strongest a "SHOULD".

-- 
Randell Jesup, Worldgate (developers of the Ojo videophone), ex-Amiga OS team
[email-masked]
"The fetters imposed on liberty at home have ever been forged out of the weapons
provided for defence against real, pretended, or imaginary dangers from abroad."
		- James Madison, 4th US president (1751-1836)