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RFC2658 - RTP Payload Format for PureVoice(tm) Audio

王朝other·作者佚名  2008-05-31
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Network Working Group K. McKay

Request for Comments: 2658 QUALCOMM Incorporated

Category: Standards Track August 1999

RTP Payload Format for PureVoice(tm) Audio

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 (1999). All Rights Reserved.

ABSTRACT

This document describes the RTP payload format for PureVoice(tm)

Audio. The packet format supports variable interleaving to redUCe

the effect of packet loss on audio quality.

1 Introduction

This document describes how compressed PureVoice audio as produced by

the Qualcomm PureVoice CODEC [1] may be formatted for use as an RTP

payload type. A method is provided to interleave the output of the

compressor to reduce quality degradation due to lost packets.

Furthermore, the sender may choose various interleave settings based

on the importance of low end-to-end delay versus greater tolerance

for lost packets.

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 [3].

2 Background

The Electronic Industries Association (EIA) & Telecommunications

Industry Association (TIA) standard IS-733 [1] defines an audio

compression algorithm for use in CDMA applications. In addition to

being the standard CODEC for all wireless CDMA terminals, the

Qualcomm PureVoice CODEC (a.k.a. Qcelp) is used in several Internet

applications most notably JFax(tm), Apple(r) QuickTime(tm), and

Eudora(r).

The Qcelp CODEC [1] compresses each 20 milliseconds of 8000 Hz, 16-

bit sampled input speech into one of four different size output

frames: Rate 1 (266 bits), Rate 1/2 (124 bits), Rate 1/4 (54 bits)

or Rate 1/8 (20 bits). The CODEC chooses the output frame rate based

on analysis of the input speech and the current operating mode

(either normal or reduced rate). For typical speech patterns, this

results in an average output of 6.8 k bits/sec for normal mode and

4.7 k bits/sec for reduced rate mode.

3 RTP/Qcelp Packet Format

The RTP timestamp is in 1/8000 of a second units. The RTP payload

data for the Qcelp CODEC has the following format:

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 [2]

+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+

RR LLL NNN

+-+-+-+-+-+-+-+-+ one or more codec data frames

....

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The RTP header has the eXPected values as described in [2]. The

extension bit is not set and this payload type never sets the marker

bit. The codec data frames are aligned on octet boundaries. When

interleaving is in use and/or multiple codec data frames are present

in a single RTP packet, the timestamp is, as always, that of the

oldest data represented in the RTP packet. The other fields have the

following meaning:

Reserved (RR): 2 bits

MUST be set to zero by sender, SHOULD be ignored by receiver.

Interleave (LLL): 3 bits

MUST have a value between 0 and 5 inclusive. The remaining two

values (6 and 7) MUST not be used by senders. If this field is

non-zero, interleaving is enabled. All receivers MUST support

interleaving. Senders MAY support interleaving. Senders that do

not support interleaving MUST set field LLL and NNN to zero.

Interleave Index (NNN): 3 bits

MUST have a value less than or equal to the value of LLL. Values

of NNN greater than the value of LLL are invalid.

3.1 Receiving Invalid Values

On receipt of an RTP packet with an invalid value of the LLL or NNN

field, the RTP packet MUST be treated as lost by the receiver for the

purpose of generating erasure frames as described in section 4.

3.2 CODEC data frame format

The output of the Qcelp CODEC must be converted into CODEC data

frames for inclusion in the RTP payload as follows:

a. Octet 0 of the CODEC data frame indicates the rate and total size

of the CODEC data frame as indicated in this table:

OCTET 0 RATE TOTAL CODEC data frame size (in octets)

-----------------------------------------------------------

0 Blank 1

1 1/8 4

2 1/4 8

3 1/2 17

4 1 35

5 reserved 8 (SHOULD be treated as a reserved value)

14 Erasure 1 (SHOULD NOT be transmitted by sender)

other n/a reserved

Receipt of a CODEC data frame with a reserved value in octet 0

MUST be considered invalid data as described in 3.1.

b. The bits as numbered in the standard [1] from highest to lowest

are packed into octets. The highest numbered bit (265 for Rate 1,

123 for Rate 1/2, 53 for Rate 1/4 and 19 for Rate 1/8) is placed

in the most significant bit (Internet bit 0) of octet 1 of the

CODEC data frame. The second highest numbered bit (264 for Rate

1, etc.) is placed in the second most significant bit (Internet

bit 1) of octet 1 of the data frame. This continues so that bit

258 from the standard Rate 1 frame is placed in the least

significant bit of octet 1. Bit 257 from the standard is placed

in the most significant bit of octet 2 and so on until bit 0 from

the standard Rate 1 frame is placed in Internet bit 1 of octet 34

of the CODEC data frame. The remaining unused bits of the last

octet of the CODEC data frame MUST be set to zero.

Here is a detail of how a Rate 1/8 frame is converted into a CODEC

data frame:

CODEC data frame

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

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

1111111111

1 (Rate 1/8) 98765432109876543210ZZZZ

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Octet 0 of the data frame has value 1 (see table above) indicating

the total data frame length (including octet 0) is 4 octets. Bits

19 through 0 from the standard Rate 1/8 frame are placed as

indicated with bits marked with "Z" being set to zero. The Rate

1, 1/4 and 1/2 standard frames are converted similarly.

3.3 Bundling CODEC data frames

As indicated in section 3, more than one CODEC data frame MAY be

included in a single RTP packet by a sender. Receivers MUST handle

bundles of up to 10 CODEC data frames in a single RTP packet.

Furthermore, senders have the following additional restrictions:

o MUST not bundle more CODEC data frames in a single RTP packet than

will fit in the MTU of the RTP transport protocol. For the

purpose of computing the maximum bundling value, all CODEC data

frames should be assumed to have the Rate 1 size.

o MUST never bundle more than 10 CODEC data frames in a single RTP

packet.

o Once beginning transmission with a given SSRC and given bundling

value, MUST NOT increase the bundling value. If the bundling

value needs to be increased, a new SSRC number MUST be used.

o MAY decrease the bundling value only between interleave groups

(see section 3.4). If the bundling value is decreased, it MUST

NOT be increased (even to the original value), although it may be

decreased again at a later time.

3.3.1 Determining the number of bundled CODEC data frames

Since no count is transmitted as part of the RTP payload and the

CODEC data frames have differing lengths, the only way to determine

how many CODEC data frames are present in the RTP packet is to

examine octet 0 of each CODEC data frame in sequence until the end of

the RTP packet is reached.

3.4 Interleaving CODEC data frames

Interleaving is meaningful only when more than one CODEC data frame

is bundled into a single RTP packet.

All receivers MUST support interleaving. Senders MAY support

interleaving.

Given a time-ordered sequence of output frames from the Qcelp CODEC

numbered 0..n, a bundling value B, and an interleave value L where n

= B * (L+1) - 1, the output frames are placed into RTP packets as

follows (the values of the fields LLL and NNN are indicated for each

RTP packet):

First RTP Packet in Interleave group:

LLL=L, NNN=0

Frame 0, Frame L+1, Frame 2(L+1), Frame 3(L+1), ... for a total of

B frames

Second RTP Packet in Interleave group:

LLL=L, NNN=1

Frame 1, Frame 1+L+1, Frame 1+2(L+1), Frame 1+3(L+1), ... for a

total of B frames

This continues to the last RTP packet in the interleave group:

L+1 RTP Packet in Interleave group:

LLL=L, NNN=L

Frame L, Frame L+L+1, Frame L+2(L+1), Frame L+3(L+1), ... for a

total of B frames

Senders MUST transmit in timestamp-increasing order. Furthermore,

within each interleave group, the RTP packets making up the

interleave group MUST be transmitted in value-increasing order of the

NNN field. While this does not guarantee reduced end-to-end delay on

the receiving end, when packets are delivered in order by the

underlying transport, delay will be reduced to the minimum possible.

Additionally, senders have the following restrictions:

o Once beginning transmission with a given SSRC and given interleave

value, MUST NOT increase the interleave value. If the interleave

value needs to be increased, a new SSRC number MUST be used.

o MAY decrease the interleave value only between interleave groups.

If the interleave value is decreased, it MUST NOT be increased

(even to the original value), although it may be decreased again

at a later time.

3.5 Finding Interleave Group Boundaries

Given an RTP packet with sequence number S, interleave value (field

LLL) L, and interleave index value (field NNN) N, the interleave

group consists of RTP packets with sequence numbers from S-N to S-N+L

inclusive. In other words, the Interleave group always consists of

L+1 RTP packets with sequential sequence numbers. The bundling value

for all RTP packets in an interleave group MUST be the same.

The receiver determines the expected bundling value for all RTP

packets in an interleave group by the number of CODEC data frames

bundled in the first RTP packet of the interleave group received.

Note that this may not be the first RTP packet of the interleave

group sent if packets are delivered out of order by the underlying

transport.

On receipt of an RTP packet in an interleave group with other than

the expected bundling value, the receiver MAY discard CODEC data

frames off the end of the RTP packet or add erasure CODEC data frames

to the end of the packet in order to manufacture a substitute packet

with the expected bundling value. The receiver MAY instead choose to

discard the whole interleave group and play silence.

3.6 Reconstructing Interleaved Audio

Given an RTP sequence number ordered set of RTP packets in an

interleave group numbered 0..L, where L is the interleave value and B

is the bundling value, and CODEC data frames within each RTP packet

that are numbered in order from first to last with the numbers 1..B,

the original, time-ordered sequence of output frames from the CODEC

may be reconstructed as follows:

First L+1 frames:

Frame 0 from packet 0 of interleave group

Frame 0 from packet 1 of interleave group

And so on up to...

Frame 0 from packet L of interleave group

Second L+1 frames:

Frame 1 from packet 0 of interleave group

Frame 1 from packet 1 of interleave group

And so on up to...

Frame 1 from packet L of interleave group

And so on up to...

BTh L+1 frames:

Frame B from packet 0 of interleave group

Frame B from packet 1 of interleave group

And so on up to...

Frame B from packet L of interleave group

3.6.1 Additional Receiver Responsibility

Assume that the receiver has begun playing frames from an interleave

group. The time has come to play frame x from packet n of the

interleave group. Further assume that packet n of the interleave

group has not been received. As described in section 4, an erasure

frame will be sent to the Qcelp CODEC.

Now, assume that packet n of the interleave group arrives before

frame x+1 of that packet is needed. Receivers SHOULD use frame x+1

of the newly received packet n rather than substituting an erasure

frame. In other words, just because packet n wasn't available the

first time it was needed to reconstruct the interleaved audio, the

receiver SHOULD NOT assume it's not available when it's subsequently

needed for interleaved audio reconstruction.

4 Handling lost RTP packets

The Qcelp CODEC supports the notion of erasure frames. These are

frames that for whatever reason are not available. When

reconstructing interleaved audio or playing back non-interleaved

audio, erasure frames MUST be fed to the Qcelp CODEC for all of the

missing packets.

Receivers MUST use the timestamp clock to determine how many CODEC

data frames are missing. Each CODEC data frame advances the

timestamp clock EXACTLY 160 counts.

Since the bundling value may vary (it can only decrease), the

timestamp clock is the only reliable way to calculate exactly how

many CODEC data frames are missing when a packet is dropped.

Specifically when reconstructing interleaved audio, a missing RTP

packet in the interleave group should be treated as containing B

erasure CODEC data frames where B is the bundling value for that

interleave group.

5 Discussion

The Qcelp CODEC interpolates the missing audio content when given an

erasure frame. However, the best quality is perceived by the

listener when erasure frames are not consecutive. This makes

interleaving desirable as it increases audio quality when dropped

packets are more likely.

On the other hand, interleaving can greatly increase the end-to-end

delay. Where an interactive session is desired, an interleave (field

LLL) value of 0 or 1 and a bundling factor of 4 or less is

recommended.

When end-to-end delay is not a concern, a bundling value of at least

4 and an interleave (field LLL) value of 4 or 5 is recommended

subject to MTU limitations.

The restrictions on senders set forth in sections 3.3 and 3.4

guarantee that after receipt of the first payload packet from the

sender, the receiver can allocate a well-known amount of buffer space

that will be sufficient for all future reception from the same SSRC

value. Less buffer space may be required at some point in the future

if the sender decreases the bundling value or interleave, but never

more buffer space. This prevents the possibility of the receiver

needing to allocate more buffer space (with the possible result that

none is available) should the bundling value or interleave value be

increased by the sender. Also, were the interleave or bundling value

to increase, the receiver could be forced to pause playback while it

receives the additional packets necessary for playback at an

increased bundling value or increased interleave.

6 Security Considerations

RTP packets using the payload format defined in this specification

are subject to the security considerations discussed in the RTP

specification [2], and any appropriate profile (for example [4]).

This implies that confidentiality of the media streams is achieved by

encryption. Because the data compression used with this payload

format is applied end-to-end, encryption may be performed after

compression so there is no conflict between the two operations.

A potential denial-of-service threat exists for data encodings using

compression techniques that have non-uniform receiver-end

computational load. The attacker can inject pathological datagrams

into the stream which are complex to decode and cause the receiver to

be overloaded. However, this encoding does not exhibit any

significant non-uniformity.

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. In a multicast

environment, pruning of specific sources may be implemented in future

versions of IGMP [5] and in multicast routing protocols to allow a

receiver to select which sources are allowed to reach it.

7 References

[1] TIA/EIA/IS-733. TR45: High Rate Speech Service Option for

Wideband Spread Spectrum Communications Systems. Available from

Global Engineering +1 800 854 7179 or +1 303 792 2181. May also

be ordered online at http://www.eia.org/eng/.

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

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

1889, January 1996.

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

Levels", BCP 14, RFC2119, March 1997.

[4] Schulzrinne, H., "RTP Profile for Audio and Video Conferences

with Minimal Control", RFC1890, January 1996.

[5] Deering, S., "Host Extensions for IP Multicasting", STD 5, RFC

1112, August 1989.

8 Author's Address

Kyle J. McKay

QUALCOMM Incorporated

5775 Morehouse Drive

San Diego, CA 92121-1714

USA

Phone: +1 858 587 1121

EMail: kylem@qualcomm.com

9 Full Copyright Statement

Copyright (C) The Internet Society (1999). All Rights Reserved.

This document and translations of it may be copied and furnished to

others, and derivative works that comment on or otherwise explain it

or assist in its implementation may be prepared, copied, published

and distributed, in whole or in part, without restriction of any

kind, provided that the above copyright notice and this paragraph are

included on all such copies and derivative works. However, this

document itself may not be modified in any way, such as by removing

the copyright notice or references to the Internet Society or other

Internet organizations, except as needed for the purpose of

developing Internet standards in which case the procedures for

copyrights defined in the Internet Standards process must be

followed, or as required to translate it into languages other than

English.

The limited permissions granted above are perpetual and will not be

revoked by the Internet Society or its successors or assigns.

This document and the information contained herein is provided on an

"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING

TASK FORCE DISCLAIMS 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.

Acknowledgement

Funding for the RFCEditor function is currently provided by the

Internet Society.

 
 
 
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