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Date: Thu, 21 Nov 96 10:19:00 PST
From: Vineet Kumar <Vineet_Kumar@ccm.jf.intel.com>
To: gary_shirk@ccm.jf.intel.com, iana@isi.edu
Subject: need payload type assigned to G.723.1 audio codec
G.723.1 is an audio codec used in ITU's H.323 and H.324. It is being
widely used for Internet Telephony. Intel's own Internet Phone product
has over 300,000 (three-hundred-thousand) downloads. Microsoft's
Netmeeting also implements H.323 with G.723.1 and is expected to reach
millions of consumers. Due to such a demand it makes a lot of sense to
have a static payload type assigned to G.723.1.
--------------------------------------------------------------------
About G.723.1
---------------
G.723.1 is specified in ITU recommendation G.723.1. Reference
implementations for G.723.1 are available as part of the CCITT/ITU-T
Software Tool Library (STL) from the ITU General Secretariat, Sales
Service, Place du Nations, CH-1211 Geneve 20, Switzerland. The library
is covered by a license.
This Recommendation specifies a coded representation that can be used
for compressing the speech or other audio signal component of
multi-media services at a very low bit rate. A G.723.1 frame can be
one of three sizes: 24 bytes (6.3 kb/s frame), 20 bytes (5.3 kb/s
frame), or 4 bytes. These 4-byte frames are called SID frames
(Silence Insertion Descriptor) and are used to specify comfort noise
parameters. There is no restriction on how 4, 20, and 24 byte frames
are intermixed. The first two bits in the frame determine the frame
boundary. It is possible to switch between the two rates at any 30 ms
frame boundary. Both (5.3 kb/s and 6.3 kb/s) rates are a mandatory
part of the encoder and decoder. This coder was optimized to represent
speech with a high quality at the above rates using a limited amount
of complexity.
Conformance to RFC 1890
------------------------
G.723.1 packetization conforms to RFC 1890 except for the
packetization interval (30 ms vs. 20 ms default):
1. The first packet of a talkspurt (first packet after a silence
period) is distinguished by setting the marker bit in the RTP data
header.
2. The sampling frequency (RTP clock frequency) is 8000 Hz.
3. The packetization interval should have a duration of 30 ms (one
frame) as opposed to the default packetization of 20 ms.
4. Codecs should be able to encode and decode several consecutive
frames within a single packet.
5. A receiver should accept packets representing between 0 and 180 ms
of audio data as opposed to the default of 0 and 200 ms.
Bibliography
-------------
1. International Telecommunication Union (ITU-T), "Recommendation
G.723.1: Dual Rate Speech Coder for Multimedia Communications
transmitting at 5.3 and 6.3 kbits/s", Geneva, Switzerland, March 1996.
(http://www.itu.ch).
2. H. Schulzrinne, "RTP Profile for Audio and Video Conferences with
Minimal Control", RFC 1890, GMD Fokus, January 1996.
Author's Address
-------------------
Vineet Kumar
Intel Corporation, JF3-212
2111 NE 25th Avenue
Hillsboro, OR 97124-5961
USA
Phone: +1 (503) 264-3439
EMail: vineet_kumar@ccm.jf.intel.com
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