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RFC4040 - RTP Payload Format for a 64 kbit/s Transparent Call

王朝other·作者佚名  2008-05-31
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Network Working Group R. Kreuter

Request for Comments: 4040 Siemens AG

Category: Standards Track April 2005

RTP Payload Format for a 64 kbit/s Transparent Call

Status of This Memo

This document specifies an Internet standards track protocol for the

Internet community, and requests discussion and suggestions for

improvements. Please refer to the current edition of the "Internet

Official Protocol Standards" (STD 1) for the standardization state

and status of this protocol. Distribution of this memo is unlimited.

Copyright Notice

Copyright (C) The Internet Society (2005).

Abstract

This document describes how to carry 64 kbit/s channel data

transparently in RTP packets, using a pseudo-codec called

"Clearmode". It also serves as registration for a related MIME type

called "audio/clearmode".

"Clearmode" is a basic feature of VoIP Media Gateways.

Table of Contents

1. IntrodUCtion.................................................. 2

2. Conventions Used in This Document............................. 2

3. 64 kbit/s Data Stream Handling and RTP Header Parameters...... 3

4. IANA Considerations........................................... 3

5. Mapping to Session Description Protocol (SDP) Parameters...... 5

6. Security Considerations....................................... 5

7. References.................................................... 6

7.1. Normative References..................................... 6

7.2. Informative References................................... 6

8. Acknowledgements.............................................. 7

1. Introduction

Voice over IP (VoIP) Media Gateways need to carry all possible data

streams generated by analog terminals or integrated services digital

network (ISDN) terminals via an IP network. Within this document a

VoIP Media Gateway is a device that converts a (digital or analog)

linear data stream to a digital packetized data stream or vice versa.

Refer to RFC 2719 [10] for an introduction into the basic

architecture of a Media Gateway based network.

Usually a VoIP Media Gateway does some processing on the data it

converts besides packetization or depacketization; i.e. echo

cancellation or dual tone multifrequency (DTMF) detection, and

especially a coding/decoding. But there is a class of data streams

that does not rely on or allow any data processing within the VoIP

Media Gateway except for packetization or depacketization. ISDN data

terminals i.e. will produce data streams that are not compatible with

a non-linear encoding as used for voice.

For such applications, there is a necessity for a transparent relay

of 64 kbit/s data streams in real-time transport protocol (RTP) [4]

packets. This mode is often referred to as "clear-channel data" or

"64 kbit/s unrestricted". No encoder/decoder is needed in that case,

but a unique RTP payload type is necessary and a related MIME type is

to be registered for signaling purposes.

Clearmode is not restricted to the examples described above. It can

be used by any application, that does not need a special

encoding/decoding for transfer via a RTP connection.

This payload format document describes a pseudo-codec called

"Clearmode", for sample oriented 64 kbit/s data streams with 8 bits

per sample. It is in accordance with RFC 2736 [1], which provides a

guideline for the specification of new RTP payload formats.

Examples for the current use of Clearmode are the transfer of "ISDN 7

kHz voice" and "ISDN data" in VoIP Media Gateways.

This document also serves as the MIME type registration according to

RFC 2045 [2] and RFC 2048 [3], which defines procedures for

registration of new MIME types within the IETF tree.

2. Conventions Used in This Document

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 RFC 2119 [8].

3. 64 kbit/s Data Stream Handling and RTP Header Parameters

Clearmode does not use any encoding or decoding. It just provides

packetization.

Clearmode assumes that the data to be handled is sample oriented with

one octet (8bits) per sample. There is no restriction on the number

of samples per packet other than the 64 kbyte limit imposed by the IP

protocol. The number of samples SHOULD be less than the path maximum

transmission unit (MTU) minus combined packet header length. If the

environment is eXPected to have tunnels or security encapsulation as

part of operation, the number of samples SHOULD be reduced to allow

for the extra header space.

The payload packetization/depacketization for Clearmode is similar to

the Pulse Code Modulation (PCMU or PCMA) handling described in RFC

3551 [5]. Each Clearmode octet SHALL be octet-aligned in an RTP

packet. The sign bit of each octet SHALL correspond to the most

significant bit of the octet in the RTP packet.

A sample rate of 8000 Hz MUST be used.

This calculates to a 64 kbit/s transmission rate per channel.

The Timestamp SHALL be set as described in RFC 3550 [4].

The marker bit is always zero. Silence suppression is not applicable

for Clearmode data streams.

The payload type is dynamically assigned and is not presented in this

document.

RTP header fields not mentioned here SHALL be used as specified in

RFC 3550 [4] and any applicable profile.

This document specifies the use of RTP over unicast and multicast UDP

as well as TCP. (This does not preclude the use of this definition

when RTP is carried by other lower-layer protocols.)

4. IANA Considerations

This document registers the following MIME suBType: audio/clearmode.

To: ietf-types@iana.org

Subject: Registration of MIME media type audio/clearmode

MIME media type name: audio

MIME subtype name: clearmode

Required parameters: none

Optional parameters: ptime, maxptime

"ptime" gives the length of time in milliseconds

represented by the media in a packet, as described in RFC

2327 [6].

"maxptime" represents the maximum amount of media, which

can be encapsulated in each packet, expressed as time in

milliseconds, as described in RFC 3267 [9].

Encoding considerations:

This type is only defined for transfer via RTP [4].

Security considerations:

See Section 6 of RFC 4040

Interoperability considerations: none

Published specification: RFC 4040

Applications, which use this media type:

Voice over IP Media Gateways, transferring "ISDN 64 kb/s

data", "ISDN 7 kHz voice", or other 64 kbit/s data streams

via an RTP connection

Note: the choice of the "audio" top-level MIME type was

made because the dominant uses of this pseudo-codec are

expected to telephony and voice-gateway-related. The

"audio" type allows the use of sharing of the port in the

SDP "m=" line with codecs such as audio/g711 [6], [7], for

one example. This sharing is an important application and

would not be possible otherwise.

Additional information: none

Intended usage: COMMON

Author/Change controller:

IETF Audio/Video transport working group

delegated from the IESG

5. Mapping to Session Description Protocol (SDP) Parameters

Parameters are mapped to SDP [6] in a standard way.

o The MIME type (audio) goes in SDP "m=" as the media name.

o The MIME subtype (clearmode) goes in SDP "a=rtpmap" as the

encoding name.

o The optional parameters "ptime" and "maxptime" go in the SDP

"a=ptime" and "a=maxptime" attributes, respectively.

An example mapping is as follows:

audio/clearmode; ptime=10

m=audio 12345 RTP/AVP 97

a=rtpmap:97 CLEARMODE/8000

a=ptime:10

Note that the payload format (encoding) names defined in the RTP

Profile are commonly shown in upper case. MIME subtypes are commonly

shown in lower case. These names are case-insensitive in both

places.

6. Security Considerations

Implementations using the payload format defined in this

specification are subject to the security considerations discussed in

the RFC 3550 [4]. The payload format described in this document does

not specify any different security services. The primary function of

this payload format is to add a transparent transport for a 64 kbit/s

data stream.

Confidentiality of the media streams is achieved by encryption, for

example by application of the Secure RTP profile [11].

As with any IP-based protocol, in some circumstances a receiver may

be overloaded simply by the receipt of too many packets, either

desired or undesired. Network-layer authentication MAY be used to

discard packets from undesired sources, but the processing cost of

the authentication itself may be too high. Overload can also occur,

if the sender chooses to use a smaller packetization period, than the

receiver can process. The ptime parameter can be used to negotiate

an appropriate packetization during session setup.

In general RTP is not an appropriate transfer protocol for reliable

octet streams. TCP is better in those cases. Besides that, packet

loss due to congestion is as much an issue for clearmode, as for

other payload formats. Refer to RFC 3551 [5], section 2, for a

discussion of this issue.

7. References

7.1. Normative References

[1] Handley, M. and C. Perkins, "Guidelines for Writers of RTP

Payload Format Specifications", BCP 36, RFC 2736, December 1999.

[2] Freed, N. and N. Borenstein, "Multipurpose Internet Mail

Extensions (MIME) Part One: Format of Internet Message Bodies",

RFC 2045, November 1996.

[3] Freed, N., Klensin, J., and J. Postel, "Multipurpose Internet

Mail Extensions (MIME) Part Four: Registration Procedures", BCP

13, RFC 2048, November 1996.

[4] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,

"RTP: A Transport Protocol for Real-Time Applications", STD 64,

RFC 3550, July 2003.

[5] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video

Conferences with Minimal Control", STD 65, RFC 3551, July 2003.

[6] Handley, M. and V. Jacobson, "SDP: Session Description

Protocol", RFC 2327, April 1998.

[7] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with

Session Description Protocol (SDP)", RFC 3264, June 2002.

[8] Bradner, S., "Key words for use in RFCs to Indicate Requirement

Levels", BCP 14, RFC 2119, March 1997.

[9] Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q. Xie, "Real-

Time Transport Protocol (RTP) Payload Format and File Storage

Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate

Wideband (AMR-WB) Audio Codecs", RFC 3267, June 2002.

7.2. Informative References

[10] Ong, L., Rytina, I., Garcia, M., Schwarzbauer, H., Coene, L.,

Lin, H., Juhasz, I., Holdrege, M., and C. Sharp, "Framework

Architecture for Signaling Transport", RFC 2719, October 1999.

[11] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.

Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC

3711, March 2004.

8. Acknowledgements

The editor would like to acknowledge the help of the IETF AVT Working

Group and, in particular the help of Colin Perkins and Magnus

Westerlund for their intensive reviews and comments.

Author's Address

Ruediger Kreuter

Siemens AG

81730 Munich, Germany

EMail: ruediger.kreuter@siemens.com

Full Copyright Statement

Copyright (C) The Internet Society (2005).

This document is subject to the rights, licenses and restrictions

contained in BCP 78, and except as set forth therein, the authors

retain all their rights.

This document and the information contained herein are provided on an

"AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS

OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND THE INTERNET

ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR IMPLIED,

INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE

INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED

WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.

Intellectual Property

The IETF takes no position regarding the validity or scope of any

Intellectual Property Rights or other rights that might be claimed to

pertain to the implementation or use of the technology described in

this document or the extent to which any license under such rights

might or might not be available; nor does it represent that it has

made any independent effort to identify any such rights. Information

on the procedures with respect to rights in RFC documents can be

found in BCP 78 and BCP 79.

Copies of IPR disclosures made to the IETF Secretariat and any

assurances of licenses to be made available, or the result of an

attempt made to obtain a general license or permission for the use of

such proprietary rights by implementers or users of this

specification can be obtained from the IETF on-line IPR repository at

http://www.ietf.org/ipr.

The IETF invites any interested party to bring to its attention any

copyrights, patents or patent applications, or other proprietary

rights that may cover technology that may be required to implement

this standard. Please address the information to the IETF at ietf-

ipr@ietf.org.

Acknowledgement

Funding for the RFC Editor function is currently provided by the

Internet Society.

 
 
 
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