< draft-ietf-avt-dsr-04.txt		draft-ietf-avt-dsr-05.txt >
			A new Request for Comments is now available in online RFC libraries.
	Internet Engineering Task Force Q. Xie (Editor)		RFC 3557
	Audio Video Transport WG Motorola, Inc.
	INTERNET-DRAFT
	Expires in six months October 4, 2002
	RTP Payload Format for ETSI ES 201 108 Distributed
	Speech Recognition Encoding
	<draft-ietf-avt-dsr-04.txt>
	Status of this Memo
	This document is an Internet-Draft and is in full conformance with
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	Abstract
	This document specifies an RTP payload format for encapsulating ETSI
	Standard ES 201 108 front-end signal processing feature streams for
	distributed speech recognition (DSR) systems.
	1. Conventions and Acronyms
	The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
	"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in
	this document are to be interpreted as described in [RFC2119].
	The following acronyms are used in this document:
	DSR - Distributed Speech Recognition
	ETSI - the European Telecommunications Standards Institute
	FP - Frame Pair
	DTX - Discontinuous Transmission
	2. Introduction
	Motivated by technology advances in the field of speech recognition,
	voice interfaces to services (such as airline information systems,
	unified messaging) are becoming more prevalent. In parallel, the
	popularity of mobile devices has also increased dramatically. However,
	the voice codecs typically employed in mobile devices were designed to
	optimize audible voice quality and not speech recognition accuracy,
	and using these codecs with speech recognizers can result in poor
	recognition performance. For systems that can be accessed from
	heterogeneous networks using multiple speech codecs, recognition
	system designers are further challenged to accommodate the
	characteristics of these differences in a robust manner. Channel
	errors and lost data packets in these networks result in further
	degradation of the speech signal.
	In traditional systems as described above, the entire speech
	recognizer lies on the server. It is forced to use incoming speech in
	whatever condition it arrives after the network decodes the vocoded
	speech. To address this problem, we use a distributed speech
	recognition (DSR) architecture. In such a system, the remote device acts as
	a thin client, also known as the front-end, in communication with a
	speech recognition server, also called a speech engine. The remote
	device processes the speech, compresses the data, and adds error
	protection to the bitstream in a manner optimal for speech
	recognition. The speech engine then uses this representation directly,
	minimizing the signal processing necessary and benefiting from
	enhanced error concealment.
	To achieve interoperability with different client devices and speech
	engines, a common format is needed. Within the "Aurora" DSR working
	group of the European Telecommunications Standards Institute (ETSI), a
	payload has been defined and was published as a standard [ES201108] in
	February 2000.
	For voice dialogues between a caller and a voice service, low latency
	is a high priority along with accurate speech recognition. While
	jitter in the speech recognizer input is not particularly important,
	many issues related to speech interaction over an IP-based connection
	are still relevant. Therefore, it is desirable to use the DSR payload
	in an RTP-based session.
	2.1. ETSI ES 201 108 DSR Front-end Codec
	The ETSI Standard ES 201 108 for DSR [ES201108] defines a signal
	processing front-end and compression scheme for speech input to a
	speech recognition system. Some relevant characteristics of this ETSI
	DSR front-end codec are summarized below.
	The coding algorithm, a standard mel-cepstral technique common to many
	speech recognition systems, supports three raw sampling rates: 8 kHz,
	11 kHz, and 16 kHz. The mel-cepstral calculation is a frame-based
	scheme that produces an output vector every 10 ms.
	After calculation of the mel-cepstral representation, the
	representation is first quantized via split-vector quantization to
	reduce the data rate of the encoded stream. Then, the quantized
	vectors from two consecutive frames are put into an FP, as
	described in more detail in Section 4.1.
	2.2 Typical Scenarios for Using DSR Payload Format
	The diagrams in Figure 1 show some typical use scenarios of the ES 201
	108 DSR RTP payload format.
	+--------+ +----------+
	|IP USER | IP/UDP/RTP/DSR |IP SPEECH |
	|TERMINAL|-------------------->| ENGINE |
	| | | |
	+--------+ +----------+
	a) IP user terminal to IP speech engine
	+--------+ DSR over +-------+ +----------+
	| Non-IP | Circuit link | | IP/UDP/RTP/DSR |IP SPEECH |
	| USER |:::::::::::::::>|GATEWAY|--------------->| ENGINE |
	|TERMINAL| ETSI payload | | | |
	+--------+ format +-------+ +----------+
	b) non-IP user terminal to IP speech engine via a gateway
	+--------+ +-------+ DSR over +----------+
	|IP USER | IP/UDP/RTP/DSR | | circuit link | Non-IP |
	|TERMINAL|----------------->|GATEWAY|::::::::::::::::>| SPEECH |
	| | | | ETSI payload | ENGINE |
	+--------+ +-------+ format +----------+
	c) IP user terminal to non-IP speech engine via a gateway
	Figure 1: Typical Scenarios for Using DSR Payload Format.
	For the different scenarios in Figure 1, the speech recognizer always
	resides in the speech engine. A DSR front-end encoder inside the User
	Terminal performs front-end speech processing and sends the resultant
	data to the speech engine in the form of "frame pairs" (FPs). Each FP
	contains two sets of encoded speech vectors representing 20ms of
	original speech.
	3. ES 201 108 DSR RTP Payload Format
	An ES 201 108 DSR RTP payload datagram consists of a standard RTP
	header [RFC1889] followed by a DSR payload. The DSR payload itself is
	formed by concatenating a series of ES 201 108 DSR FPs (defined in
	Section 4).
	FPs are always packed bit-contiguously into the payload octets
	beginning with the most significant bit. For ES 201 108 front-end, the
	size of each FP is 96 bits or 12 octets (see Sections 4.1 and
	4.2). This ensures that a DSR payload will always end on an octet
	boundary.
	The following example shows a DSR RTP datagram carrying a DSR payload
	containing three 96-bit-long FPs (bit 0 is the MSB):
	0 1 2 3
	0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
	+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
	\ \
	/ RTP header in [RFC1889] /
	\ \
	+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
	| |
	+ +
	| FP #1 (96 bits) |
	+ +
	| |
	+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
	| |
	+ +
	| FP #2 (96 bits) |
	+ +
	| |
	+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
	| |
	+ +
	| FP #3 (96 bits) |
	+ +
	| |
	+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
	Figure 2. An example of an ES 201 108 DSR RTP payload.
	3.1. Consideration on Number of FPs in Each RTP Packet
	The number of FPs per payload packet should be determined by the
	latency and bandwidth requirements of the DSR application using this
	payload format. In particular, using a smaller number of FPs per
	payload packet in a session will result in lowered bandwidth
	efficiency due to the RTP/UDP/IP header overhead, while using a larger
	number of FPs per packet will cause longer end-to-end delay and hence
	increased recognition latency. Furthermore, carrying a larger number of
	FPs per packet will increase the possibility of catastrophic packet
	loss; the loss of a large number of consecutive FPs is a situation
	most speech recognizers have difficulty dealing with.
	It is therefore RECOMMENDED that the number of FPs per DSR payload
	packet be minimized, subject to meeting the application's requirements
	on network bandwidth efficiency. RTP header compression techniques,
	such as those defined in [RFC2508] and [RFC3095], should be considered
	to improve network bandwidth efficiency.
	3.2. Support for Discontinuous Transmission
	The DSR RTP payloads may be used to support discontinuous transmission
	(DTX) of speech, which allows that DSR FPs are sent only when speech
	has been detected at the terminal equipment.
	In DTX a set of DSR frames coding an unbroken speech segment
	transmitted from the terminal to the server is called a transmission
	segment. A DSR frame inside such a transmission segment can be
	either a speech frame or a non-speech frame, depending on the nature
	of the section of the speech signal it represents.
	The end of a transmission segment is determined at the sending end
	equipment when the number of consecutive non-speech frames exceeds a
	pre-set threshold, called the hangover time. A typical value used for
	the hangover time is 1.5 seconds.
	After all FPs in a transmission segment are sent, the front-end SHOULD
	indicate the end of the current transmission segment by sending one or
	more Null FPs (defined in Section 4.2).
	4. Frame Pair Formats
	4.1. Format of Speech and Non-speech FPs
	The following mel-cepstral frame MUST be used, as defined in
	[ES201108]:
	0 1 2 3
	0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
	+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
	| idx(0,1) | idx(2,3) | idx(4,5) | idx(6,7) | idx(8,9) |idx
	+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
	(10,11) | idx(12,13) |
	+-+-+-+-+-+-+-+-+-+-+-+-+
	The length of a frame is 44 bits representing 10ms of voice.
	As defined in [ES201108], pairs of the quantized 10ms mel-cepstral
	frames MUST be grouped together and protected with a 4-bit CRC,
	forming a 92-bit long FP:
	0 1 2 3
	0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
	+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
	| Frame #1 (44 bits) |
	+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
	| | Frame #2 (44 bits) |
	+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+
	| | CRC |0|0|0|0|
	+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
	Therefore, each FP represents 20ms of original speech. Note, as shown
	above, each FP MUST be padded with 4 zeros to the end in order to make
	it aligned to the 32-bit word boundary. This makes the size of an FP
	96 bits, or 12 octets. Note, this padding is separate from padding
	indicated by the P bit in the RTP header.
	The 4-bit CRC MUST be calculated using the formula defined in 6.2.4 in
	[ES201108].
	4.2. Format of Null FP
	A Null FP for the ES 201 108 front-end codec is defined by setting the
	content of the first and second frame in the FP to null (i.e., filling
	the first 88 bits of the FP with 0's). The 4-bit CRC MUST be
	calculated the same way as described in 6.2.4 in [ES201108], and 4
	zeros MUST be padded to the end of the Null FP to made it 32-bit word
	aligned.
	4.3. RTP header usage
	The format of the RTP header is specified in [RFC1889]. This payload
	format uses the fields of the header in a manner consistent with that
	specification.
	The RTP timestamp corresponds to the sampling instant of the first
	sample encoded for the first FP in the packet. The timestamp
	clock frequency is the same as the sampling frequency, so the
	timestamp unit is in samples.
	As defined by ES 201 108 front-end codec, the duration of one FP is 20
	ms, corresponding to 160, 220, or 320 encoded samples with sampling
	rate of 8, 11, or 16 kHz being used at the front-end, respectively.
	Thus, the timestamp is increased by 160, 220, or 320 for each
	consecutive FP, respectively.
	The DSR payload for ES 201 108 front-end codes is always an integral
	number of octets. If additional padding is required for some other
	purpose, then the P bit in the RTP in the header may be set and
	padding appended as specified in [RFC1889].
	The RTP header marker bit (M) should be set following the general
	rules defined in [RFC1890new].
	The assignment of an RTP payload type for this new packet format is
	outside the scope of this document, and will not be specified here. It
	is expected that the RTP profile under which this payload format is
	being used will assign a payload type for this encoding or specify
	that the payload type is to be bound dynamically.
	5. DSR MIME Type Registration
	Media Type name: audio
	Media subtype name: dsr-es201108
	Required parameters: none
	Optional parameters for RTP mode:
	rate: Indicates the sample rate of the speech. Valid values include:
	8000, 11000, and 16000. If this parameter is not present,
	8000 sample rate is assumed.
	maxptime: The maximum amount of media which can be encapsulated in
	each packet, expressed as time in milliseconds. The time shall
	be calculated as the sum of the time the media present in the
	packet represents. The time SHOULD be a multiple of the frame
	pair size (i.e., one FP <-> 20ms).
	If this parameter is not present, maxptime is assumed to be
	80ms.
	Note, since the performance of most speech recognizers are
	extremely sensitive to consecutive FP losses, if the user of
	the payload format expects a high packet loss ratio for the
	session, it MAY consider to explicitly choose a maxptime
	value for the session that is shorter than the default value.
	ptime: see RFC2327 [RFC2327].
	Encoding considerations : This type is defined for transfer via RTP
	[RFC1889] as described in Sections 3 and 4 of RFC XXXX.
	Security considerations : See Section 6 of RFC XXXX.
	Person & email address to contact for further information:
	Qiaobing.Xie@motorola.com
	Intended usage: COMMON. It is expected that many VoIP applications
	(as well as mobile applications) will use this type.
	Author/Change controller:
	Qiaobing.Xie@motorola.com
	IETF Audio/Video transport working group
	5.1. Mapping MIME Parameters into SDP
	The information carried in the MIME media type specification has a
	specific mapping to fields in the Session Description Protocol (SDP)
	[RFC2327], which is commonly used to describe RTP sessions. When SDP is
	used to specify sessions employing ES 201 018 DSR codec, the
	mapping is as follows:
	o The MIME type ("audio") goes in SDP "m=" as the media name.
	o The MIME subtype ("dsr-es201108") goes in SDP "a=rtpmap" as the
	encoding name.
	o The optional parameter "rate" also goes in "a=rtpmap" as clock
	rate.
	o The optional parameters "ptime" and "maxptime" go in the SDP
	"a=ptime" and "a=maxptime" attributes, respectively.
	Example of usage of ES 201 108 DSR:
	m=audio 49120 RTP/AVP 101
	a=rtpmap:101 dsr-es201108/8000
	a=maxptime:40
	6. Security Considerations
	Implementations using the payload defined in this specification are
	subject to the security considerations discussed in the RTP
	specification [RFC1889] and the RTP profile [RFC1890new]. This payload
	does not specify any different security services.
	7. Contributors
	The following individuals contributed to the design of this payload
	format and the writing of this document: Q. Xie (Motorola), D. Pearce
	(Motorola), S. Balasuriya (Motorola), Y. Kim (VerbalTek), S. H. Maes
	(IBM), and, Hari Garudadri (Qualcomm).
	8. Acknowledgments
	The design presented here benefits greatly from an earlier work on DSR
	RTP payload design by Jeff Meunier and Priscilla Walther. The authors
	also wish to thank Brian Eberman, John Lazzaro, Magnus Westerlund,
	Rainu Pierce, Priscilla Walther, and others for their review and
	valuable comments on this document.
	9. References
	[ES201108] European Telecommunications Standards Institute (ETSI)
	Standard ES 201 108, "Speech Processing, Transmission and Quality
	Aspects (STQ); Distributed Speech Recognition; Front-end Feature
	Extraction Algorithm; Compression Algorithms," Ver. 1.1.2, April
	11, 2000. http://webapp.etsi.org/pda/home.asp?wki_id=9948
	[RFC1889] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson,
	"RTP: A transport protocol for real-time applications," Internet
	Draft, Internet Engineering Task Force, Feb. 1999 Work in progress,
	revision to RFC 1889.
	[RFC2026] Bradner, S., "The Internet Standards Process -- Revision 3",
	BCP 9, RFC 2026, October 1996.
	[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
	Requirement Levels", BCP 14, RFC 2119, March 1997
	[RFC2327] M. Handley and V. Jacobson, "SDP: Session Description
	Protocol", IETF RFC 2327, April 1998
	9.1. Informative References		Title: RTP Payload Format for European Telecommunications
			Standards Institute (ETSI) European Standard
			ES 201 108 Distributed Speech Recognition Encoding
			Author(s): Q. Xie, Ed.
			Status: Standards Track
			Date: July 2003
			Mailbox: Qiaobing.Xie@motorola.com
			Pages: 15
			Characters: 29692
			Updates/Obsoletes/SeeAlso: None
	[RFC1890new] H. Schulzrinne and S. Casner, "RTP Profile for Audio and		I-D Tag: draft-ietf-avt-dsr-05.txt
	Video Conferences with Minimal Control," Internet Draft
	draft-ietf-avt-profile-new-12.txt, Work in Progress November 21,
	2000, revision to RFC 1890.
	[RFC2508] S. Casner and V. Jacobson, "Compressing IP/UDP/RTP Headers		URL: ftp://ftp.rfc-editor.org/in-notes/rfc3557.txt
	for Low-Speed Serial Links," RFC 2508, February 1999.
	[RFC3095] C. Bormann, C. Burmeister, M. Degermark, H. Fukushima,		This document specifies an RTP payload format for encapsulating
	H. Hannu, L-E. Jonsson, R. Hakenberg, T. Koren, K. Le, Z. Liu,		European Telecommunications Standards Institute (ETSI) European
	A. Martensson, A. Miyazaki, K. Svanbro, T. Wiebke, T. Yoshimura,		Standard (ES) 201 108 front-end signal processing feature streams
	and H. Zheng, "RObust Header Compression (ROHC): Framework and four		for distributed speech recognition (DSR) systems.
	profiles: RTP, UDP, ESP, and uncompressed," IETF RFC 3095, July
	2001.
	10. Author's Addresses		This document is a product of the Audio/Video Transport Working Group
			of the IETF.
	Qiaobing Xie Tel: +1-847-632-3028		This is now a Proposed Standard Protocol.
	Motorola, Inc. EMail: Qiaobing.Xie@motorola.com
	1501 W. Shure Drive, 2-F9
	Arlington Heights, IL 60004, USA
	David Pearce Tel: +44 (0)1256 484 436		This document specifies an Internet standards track protocol for
	Motorola Labs EMail: bdp003@motorola.com		the Internet community, and requests discussion and suggestions
	UK Research Laboratory		for improvements. Please refer to the current edition of the
	Jays Close		"Internet Official Protocol Standards" (STD 1) for the
	Viables Industrial Estate		standardization state and status of this protocol. Distribution
	Basingstoke, HANTS, RG22 4PD		of this memo is unlimited.
	Senaka Balasuriya Tel: +1-630-353-8347		This announcement is sent to the IETF list and the RFC-DIST list.
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	Yoon Kim Tel: +1-408-768-4974		Details on obtaining RFCs via FTP or EMAIL may be obtained by sending
	VerbalTek, Inc. EMail: yoonie@verbaltek.com		an EMAIL message to rfc-info@RFC-EDITOR.ORG with the message body
	2921 Copper Rd.		help: ways_to_get_rfcs. For example:
	Santa Clara, CA 95051
	Stephane H. Maes Tel: +1-914-945-2908		To: rfc-info@RFC-EDITOR.ORG
	IBM EMail: smaes@us.ibm.com		Subject: getting rfcs
	TJ Watson Research Center
	P.O. Box 218,
	Yorktown Heights, NY 10598, USA.
	Hari Garudadri Tel: +1-858-651-6383		help: ways_to_get_rfcs
	Qualcomm Inc. EMail: hgarudad@qualcomm.com
	5775, Morehouse Dr.
	San Diego, CA 92121-1714, USA
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