Internet Draft Adam H. Li
draft-ietf-avt-evrc-08.txt UCLA
October 10, 2001 Editor
Expires: April 10, 2002
An RTP Payload Format for EVRC Speech
STATUS OF THIS MEMO
This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC 2026.
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ABSTRACT
This document describes the RTP payload format for Enhanced Variable
Rate Codec (EVRC) Speech. The packet format supports various formats
for different application scenarios. A bundled/interleaved format is
included to reduce the effect of packet loss on Speech quality. A
non-bundled format is also supported for conversational applications.
Table of Contents
1. Introduction ................................................... 2
2. Background ..................................................... 2
3. RTP/EVRC Packet Format ......................................... 3
3.1. Type 1 RTP/EVRC Packet Format ................................ 3
3.2. Type 2 RTP/EVRC Packet Format ................................ 4
3.3. Detection Between the Type 1 and Type 2 Packets .............. 4
4. Packet Table of Content Entries and Codec Data Frame Format .... 5
4.1. Packet Table of Content entries .............................. 5
4.2. The Codec Data Frame ......................................... 6
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5. Interleaving Codec Data Frames in Type 1 Packets ............... 7
5.1. Finding Interleave Group Boundaries .......................... 8
5.2. Reconstructing Interleaved Speech ............................ 8
5.3. Receiving Invalid Interleaving Values ........................ 9
5.4. Additional Receiver Responsibilities ......................... 9
6. Bundling Codec Data Frames in Type 1 Packets ................... 9
7. Handling Lost RTP Packets ..................................... 10
8. Implementation Issues ......................................... 10
8.1. Interleaving Length ......................................... 10
8.2. Signaling of Reduce Rate .................................... 11
9. IANA Considerations ........................................... 11
9.1 Storage Mode ................................................. 11
9.2 EVRC MIME Registration ....................................... 12
10. Mapping to SDP Parameters .................................... 13
11. Security Considerations ...................................... 13
12. Acknowledgements ............................................. 14
13. References ................................................... 14
14. AuthorsË Address ............................................. 14
1. Introduction
This document describes how compressed EVRC speech as produced by the
EVRC codec [1] may be formatted for use as an RTP payload type.
Methods are provided to packetize the codec data frames into RTP
packets, in interleaved/bundled and zero-header formats. The sender
may choose among various formats the best solutions for different
application scenarios based on the network condition, bandwidth
restriction, delay requirements, and packet-loss tolerance.
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 [2].
2. Background
The Electronic Industries Association (EIA) & Telecommunications
Industry Association (TIA) standard IS-127 [1] defines a speech
compression algorithm for use in cdma2000 applications. IS-127, or
EVRC, is the emerging speech codec standard for cdma2000.
The EVRC codec [1] compresses each 20 milliseconds of 8000 Hz, 16-
bit sampled input speech into one of three different size output
frames: Rate 1 (171 bits), Rate 1/2 (80 bits), or Rate 1/8 (16 bits).
The codec chooses the output frame rate based on analysis of the
input speech and the current operating mode (either normal or one of
several reduced rates). For typical speech patterns, this results in
an average output of 4.2 kilobits/second for normal mode and lower
for reduced rate modes.
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3. RTP/EVRC Packet Format
The RTP timestamp is in 1/8000 of a second units. The RTP payload
data for the EVRC codec MUST be transmitted in packets of one of the
following two types.
3.1 Type 1 RTP/EVRC Packet Format
This format is intended for the situation where the sender and the
receiver use interleaving/bundling to send one or more codec data
frames per packet. The RTP packet for this format is as follows:
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 [3] |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| RR| LLL | NNN | |
+-+-+-+-+-+-+-+-+ one or more ToC entries +-------------+
| | |
+-------------------------------------------------+ |
| |
| one or more codec data frames |
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The RTP header has the expected values as described in [3]. The M bit
should be set as specified in the applicable RTP profile, for
example, RFC 1890 [4]. Note that RFC 1890 [4] specifies that if the
sender does not suppress silence (i.e., sends a frame on every 20
millisecond interval), the M bit will always be zero. When 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 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 for a particular class of
applications will assign a payload type for this encoding, or if that
is not done, then a payload type in the dynamic range shall be chosen
by the sender.
The first octet of a Type 1 format packet is the Interleave Byte.
The bits within the Interleave Byte are specified as follows:
Reserved (RR): 2 bits
Reserved bits. MUST be set to zero by sender, SHOULD be ignored
by receiver.
Interleave Length (LLL): 3 bits
Indicates the length of interleave. MUST have a value between 0
and 7 inclusive (where a value 0 indicates bundling, a special
case of interleaving). See Section 5 and Section 6 for more
detailed discussion.
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Interleave Index (NNN): 3 bits
Indicates the index within a interleaving group. MUST have a value
less than or equal to the value of LLL. Values of NNN greater
than the value of LLL are invalid.
The Table of Content field (ToC) contains the index(es) for the codec
data frame(s) in the packet. There is one entry for each codec data
frame. The detailed formats of the ToC field and codec data frame are
specified in Section 4.
More than one codec data frames MAY be included in a single Type 1
RTP/EVRC packet by a sender. Multiple data frames may be included
within a Type 1 packet with interleaving/bundling format as described
in Section 5 and Section 6.
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 a Type 1 RTP/EVRC packet is
to examine the ToC fields of the packet.
3.2 Type 2 RTP/EVRC Packet Format
The Type 2 RTP/EVRC Packet Format is designed for maximum efficiency
and low latency in transmission of the EVRC codec data. Exactly one
codec data frame MUST be sent in each Type 2 RTP/EVRC packet. There
MUST NOT be ToC field preceding the codec data. The EVRC codec rate
for the data frame can be found out at the receiver from the length
of the codec data frame, since there is only one codec data frame in
each Type 2 packet. The Reduce Rate Signal (See Section 4.1) can not
be send in-bound with the Type 2 packets because of the lacking of
the ToC field in this type.
Use of the RTP header fields for Type 2 RTP/EVRC Packet Format is the
same as described in Section 3.1 for Type 1 RTP/EVRC Packet Format.
The detailed formats of the codec data frame are specified in Section
4.
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 [3] |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| |
+ ONLY one codec data frame +-+-+-+-+-+-+-+-+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
3.3 Detection Between the Type 1 and Type 2 Packets
All receivers MUST be able to process both types of packets. The
sender MAY choose to use one or both types of packets.
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The packets of the two types can be distinguished by using different
payload type value for the two packet types at the sender and
checking the payload type field in the RTP header at the receiver.
The association of payload type number with the packet type is done
out-of-band, for example by SDP during the setup of a session.
4. Packet Table of Content Entries and Codec Data Frame Format
4.1 Packet Table of Content entries
For each of the codec data frames in Type 1 packets, there is a
corresponding Table of Content (ToC) entry. The ToC entry includes
flags that indicates if there are more entries following the current
one, if rate reduction on the reverse direction is desired, and the
rate of the corresponding codec frame. Type 2 packets MUST NOT have
the ToC field, and there is always only one codec data frame in each
Type 2 packet.
Each ToC entry is one octet in size. The format of the octet is
indicated below:
0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
|F|D| frm type |
+-+-+-+-+-+-+-+-+
Further Entry Indication (F): 1 bit
Indicates if there are more ToC entries following the current ToC
entry. F = 1 indicates the next octet is another ToC entry. F = 0
indicates that the current entry is the final ToC entry.
Reduce Rate (D): 1 bit
Setting the 'D' bit indicates that the sender is requesting a
reduced codec rate for the reverse direction. When the 'D' bit is
not set, the sender is requesting that the codec resume normal
operation. In the case of packet loss, the codec SHOULD continue
to operate in the mode indicated by the last codec frame
received. Receivers are NOT REQUIRED to respond to the Reduce
Rate signal. (See more discussion in Section 8.2).
Frame Type: 6 bits
The frame type values and size of the associated codec data frame
are described in the table below:
Value Rate Total codec data frame size (in octets)
---------------------------------------------------------
0 Blank 0
1 1/8 2
3 1/2 10
4 1 22
14 Erasure 0 (SHOULD NOT be transmitted by sender)
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All values not listed in the above table MUST be considered
reserved. Receipt of a ToC entry with a reserved value in Frame
Type MUST be considered invalid data.
4.2 The Codec Data Frame
The output of the EVRC codec MUST be converted into codec data frames
for inclusion in the RTP payload as specified below.
The codec output data bits as numbered in the standard [1] are packed
into octets. The lowest numbered bit (bit 1 for Rate 1, Rate 1/2 and
Rate 1/8) is placed in the most significant bit (internet bit 0) of
octet 1 of the codec data frame, the second lowest bit is placed in
the second most significant bit of the first octet, the third lowest
in the third most significant bit of the first octet, and so on.
This continues until all of the bits have been placed in the codec
data frame. The remaining unused bits of the last octet of the codec
data frame MUST be set to zero. Note that this is only applicable to
Rate 1 frames (171 bits) as the Rate 1/2 (80 bits) and Rate 1/8
frames (16 bits) fit exactly into a whole number of octets.
Following is a detailed listing showing a Rate 1 EVRC codec output
frame converted into a codec data frame:
The codec data frame for a EVRC codec Rate 1 frame is 22-byte long.
Bits 1 through 171 from the EVRC codec Rate 1 frame are placed as
indicated, with bits marked with "Z" set to zero. EVRC codec Rate 1/8
and Rate 1/2 frames are converted similarly, but do not require zero
padding because they align on octet boundaries.
Rate 1 codec data frame (bytes 0 - 3)
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|
|0|0|0|0|0|0|0|0|0|1|1|1|1|1|1|1|1|1|1|2|2|2|2|2|2|2|2|2|2|3|3|3|
|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|2|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Rate 1 codec data frame (bytes 19 - 21)
1 1 1 1
4 5 6 7
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 2 3 4 5
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1| | | | | |
|4|4|4|4|4|5|5|5|5|5|5|5|5|5|5|6|6|6|6|6|6|6|6|6|6|7|7|Z|Z|Z|Z|Z|
|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| | | | | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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5. Interleaving Codec Data Frames in Type 1 Packets
As indicated in Section 3.1, more than one codec data frame MAY be
included in a single Type 1 packet by a sender. This is accomplished
by interleaving/bundling. Interleaving/bundling of codec data frames
is signaled by setting the LLL value in the Interleaving Byte to a
value between 0 and 7 inclusive.
The special case with the LLL value set to 0 is a reduced and
simplified case of interleaving. This is sometimes called bundling,
because multiple consecutive codec data frames are included in one
RTP packet in this case. The discussions on general interleaving
apply to the bundling case with reduced complexity. The bundling case
is discussed in detail in Section 6.
Senders MAY support interleaving/bundling. All receivers MUST support
interleaving/bundling.
Given a time-ordered sequence of output frames from the EVRC codec
numbered 0..n, a bundling value B, and an interleave length 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.
Receivers MAY signal the maximum number of codec data frames (i.e.,
the maximum acceptable bundling value B) they can handle in a single
RTP packet using the OPTIONAL maxptime RTP mode parameter identified
in Section 9.2.
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Receivers MAY signal the maximum interleave length (i.e., the maximum
acceptable LLL value in the Interleaving Byte) they will accept using
the OPTIONAL maxinterleave RTP mode parameter identified in Section
9.2.
Additionally, senders have the following restrictions:
o MUST NOT bundle more codec data frames in a single RTP packet than
indicated by maxptime (see Section 9.2) if it is signaled.
o SHOULD NOT bundle more codec data frames in a single RTP packet
than will fit in the MTU of the underlying network. For the
purpose of computing the maximum bundling value, all codec data
frames MUST be assumed to have the Rate 1 size.
o Once beginning a session with a given maximum interleaving value
set by maxinterleave in Section 9.2, MUST NOT increase the
interleaving value (LLL) exceeding the maximum interleaving value
that is signaled.
o MAY change the interleaving value only between interleave groups.
5.1 Finding Interleave Group Boundaries
Given an RTP packet with sequence number S, interleave length (field
LLL) L, interleave index value (field NNN) N, and bundling value B,
the interleave group consists of RTP packets with sequence numbers
from S-N to S-N+L inclusive. (The sequence numbers used here are for
illustrative purposes. When wrapping around happens, the sequence
numbers need to be adjusted accordingly). 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.
5.2 Reconstructing Interleaved Speech
Given an RTP sequence number ordered set of RTP packets in an
interleave group numbered 0..L, where L is the interleave length and
B is the bundling value, and codec data frames within each RTP packet
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that are numbered in order from first to last with the numbers 1..B,
the original, time-ordered sequence of output frames from the EVRC
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
5.3 Receiving Invalid Interleaving 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 7.
5.4 Additional Receiver Responsibilities
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 7, an erasure
frame will be sent to the receiving EVRC 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 was not available the
first time it was needed to reconstruct the interleaved speech, the
receiver SHOULD NOT assume it is not available when it is
subsequently needed for interleaved speech reconstruction.
6. Bundling Codec Data Frames in Type 1 Packets
As discussed in Section 5, the bundling of codec data frames is a
special reduced case of interleaving with LLL value in the Interleave
Byte set to 0.
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Bundling codec data frames indicates multiple data frames are
included consecutively in a packet, because the interleaving length
(LLL) is 0. The interleaving group is thus reduced to a single RTP
packet, and the reconstruction of the code data frames from RTP
packets becomes a much simpler process.
Furthermore, the additional restriction on the senders are reduced
to:
o MUST NOT bundle more codec data frames in a single RTP packet than
indicated by maxptime (see Section 9.2) if it is signaled.
o SHOULD NOT bundle more codec data frames in a single RTP packet
than will fit in the MTU of the underlying network. For the
purpose of computing the maximum bundling value, all codec data
frames MUST be assumed to have the Rate 1 size.
7. Handling Lost RTP Packets
The EVRC codec supports the notion of erasure frames. These are
frames that for whatever reason are not available. When
reconstructing or playing back speech, erasure frames MUST be fed to
the receiving EVRC 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 interleaving length/bundling value may vary, 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 speech, a missing RTP
packet in the interleave group MUST be treated as containing B
erasure codec data frames where B is the bundling value for that
interleave group.
8. Implementation Issues
8.1 Interleaving Length
The EVRC codec interpolates the missing speech 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 speech quality when packet loss may occur.
On the other hand, interleaving can greatly increase the end-to-end
delay. Where an interactive session is desired, either Type 1 with
interleaving length 0 or Type 2 RTP payload types are RECOMMENDED.
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When end-to-end delay is not a concern, an interleaving length (field
LLL) of 4 or 5 is RECOMMENDED.
The parameters maxptime and maxinterleave at the initial setup of the
session guarantees that the receiver can allocate a well-known amount
of buffer space at the beginning of the session that will be
sufficient for all future reception in that session. Less buffer
space may be required at some point in the future if the sender
decreases the bundling value or interleaving length, 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).
8.2 Signaling of Reduce Rate
The Reduce Rate signal requests a reduction of the codec rate on the
reverse direction. It is NOT REQUIRED that all implementations react
to the Reduce Rate signal. If an implementation does react to the
Reduce Rate signal, it MUST be able to process/react to the D bit in
Type 1 packets. The Reduce Rate signal SHOULD only be used in one-to-
one sessions. In multiparty sessions, all the received Reduce Rate
signals MUST be ignored.
In addition, the Reduce Rate signal MAY also be sent through non-RTP
means, which is out of the scope of this specification.
9. IANA Considerations
One new MIME sub-type as described in this section is to be
registered.
The MIME-name for the EVRC codec is allocated from the IETF tree
since EVRC is expected to be a widely used codec for Voice-over-IP
applications.
The RTP mode has been described in the previous sections.
9.1 Storage Mode
The storage mode is used for storing speech frames, e.g. as a file or
e-mail attachment.
The file begins with a magic number to identify that it is an EVRC
file. The magic number for EVRC corresponds to the ASCII character
string "#!EVRC\n", i.e., "0x23 0x21 0x45 0x56 0x52 0x43 0x0A" in
network byte order.
The codec data frames are stored in consecutive order, with a single
TOC entry field (1 octet) prefixing each codec data frame. The F bit
and the D bit in the ToC entry field SHOULD be set to 0 and MUST be
ignored when processing speech data from storage mode.
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Speech frames lost in transmission and non-received frames MUST be
stored as erasure frames (frame type 14, see definition in Section
4.1) to maintain synchronization with the original media.
9.2 EVRC MIME Registration
Media Type Name: audio
Media Subtype Name: EVRC
Required Parameters:
ptype: Indicates the Type of the RTP/EVRC packets. The valid
values are 1 (Type 1) or 2 (Type 2).
Optional parameters for RTP mode:
ptime: Defined as usual for RTP audio [5].
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
duration of a single codec data frame (20 msec). If not
signaled, the default maxptime value SHALL be 200
milliseconds.
maxinterleave: Maximum number for interleaving length (field LLL
in the Interleaving Byte). The interleaving lengths used in
the entire session MUST NOT exceed this maximum value. If not
signaled, the maxinterleave length SHALL be 5.
Optional parameters for storage mode: none
Encoding considerations for RTP mode: see Section 5 and Section 6 of
RFC xxxx.
Encoding considerations for storage mode: see Section 9.1 of RFC
xxxx.
Security considerations: see Section 11 "Security Considerations" of
RFC xxxx.
Public specification: RFC xxxx.
Additional information for storage mode:
Magic number: #!EVRC\n
File extensions: evc, EVC
Macintosh file type code: none
Object identifier or OID: none
Intended usage: COMMON. It is expected that many VoIP applications
(as well as mobile applications) will use this type.
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Person & email address to contact for further information:
adamli@icsl.ucla.edu
Author/Change controller:
adamli@icsl.ucla.edu
IETF Audio/Video transport working group
10. Mapping to SDP Parameters
Please note that this section applies to the RTP mode only.
Parameters are mapped to SDP [5] as usual.
Example usage in SDP:
m = audio 49120 RTP/AVP 97
a = rtpmap:97 EVRC
a = fmtp:97 ptype=1; maxinterleave=2
a = maxptime:80
11. Security Considerations
RTP packets using the payload format defined in this specification
are subject to the security considerations discussed in the RTP
specification [3], 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 encoding 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
become 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 [6] and in multicast routing protocols to
allow a receiver to select which sources are allowed to reach it.
Interleaving MAY affect encryption. Depending on the used encryption
scheme there MAY be restrictions on for example the time when keys
can be changed.
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12. Acknowledgements
The editor thanks the following authors for contributions to this
document: J. D. Villasenor, D.S. Park, J.H. Park, K. Miller, S. C.
Greer, D. Leon, N. Leung, K. J. McKay, M. Lioy, T. Hiller, P. J.
McCann, M. D. Turner, A. Rajkumar, D. Gal, M. Westerlund, L.-E.
Jonsson, G. Sherwood, and T. Zeng.
13. References
[1] TIA/EIA/IS-127, "Enhanced Variable Rate Codec, Speech Service
Option 3 for Wideband Spread Spectrum Digital Systems", January
1997.
[2] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
[3] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson,
"RTP: A Transport Protocol for Real-Time Applications", RFC
1889, January 1996.
[4] Schulzrinne, H., "RTP Profile for Audio and Video Conferences
with Minimal Control", RFC 1890, January 1996.
[5] M. Handley and V. Jacobson, "SDP: Session Description Protocol",
RFC 2327, April 1998.
[6] Deering, S., "Host Extensions for IP Multicasting", STD 5, RFC
1112, August 1989.
14. Authors' Address
Adam H. Li
Image Communication Lab
Electrical Engineering Department
University of California
Los Angeles, CA 90095
USA
Phone: +1 310 825 5178
Email: adamli@icsl.ucla.edu
John D. Villasenor
Image Communication Lab
Electrical Engineering Department
University of California
Los Angeles, CA 90095
USA
Phone: +1 310 825 0228
Email: villa@icsl.ucla.edu
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INTERNET-DRAFT An RTP Payload Format for EVRC Speech October 10, 2001
Dong-Seek Park
Samsung Electronics
Suwon, Kyungki 442-742
Korea
Phone: +82 31 200 3674
Email: dspark@samsung.com
Jeong-Hoon Park
Samsung Electronics
Suwon, Kyungki 442-742
Korea
Phone: +82 31 200 3747
Email: dspark@samsung.com
Keith Miller
Nokia
6000 Connection Drive
Irving, Texas 75039
USA
Phone: +1 972 894 4296
Email: keith.miller@nokia.com
S. Craig Greer
Nokia
6000 Connection Drive
Irving, Texas 75039
USA
Phone: +1 972 894 4867
Email: craig.greer@nokia.com
David Leon
Nokia
6000 Connection Drive
Irving, Texas 75039
USA
Phone: +1 972 374 1860
Email: david.leon@nokia.com
Marcello Lioy
QUALCOMM, Incorporated
5775 Morehouse Drive
San Diego, CA 92121
USA
Phone: +1 858 651 8220
Email: mlioy@qualcomm.com
Nikolai Leung
QUALCOMM, Incorporated
7710 Takoma Ave.
Takoma Park, MD 20912
USA
Phone: +1 703 346 8351
Email: nleung@qualcomm.com
Adam H. Li [Page 15]
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Kyle J. McKay
QUALCOMM, Incorporated
5775 Morehouse Drive
San Diego, CA 92121-1714
USA
Phone: +1 858 587 1121
EMail: kylem@qualcomm.com
Tom Hiller
Lucent Technologies
263 Shuman Drive, Room 2F-218
Naperville, IL 60137
USA
Phone: +1 630 979 7673
Email: tom.hiller@lucent.com
Peter J. McCann
Lucent Technologies
263 Shuman Drive, Room 2Z-305
Naperville, IL 60137
USA
Phone: +1 630 713 9359
Email: mccap@lucent.com
Michael D. Turner
Lucent Technologies
67 Whippany Rd, Room 2A-203
Whippany, NJ 07981
USA
Phone: +1 973 386 3579
Email: mdturner@lucent.com
Ajay Rajkumar
Lucent Technologies
67 Whippany Rd, Room 1A-235
Whippany, NJ 07981
USA
Phone: +1 973 386 5249
Email: ajayrajkumar@lucent.com
Dan Gal
Lucent Technologies
67 Whippany Rd
Whippany, NJ 07981
USA
Phone: +1 973 428 7734
Email: dgal@lucent.com
Adam H. Li [Page 16]
INTERNET-DRAFT An RTP Payload Format for EVRC Speech October 10, 2001
Magnus Westerlund
Ericsson Radio Systems AB
Torshamnsgatan 23
SE-164 80 Stockholm
Sweden
Phone: +46 8 4048287
Email: magnus.westerlund@ericsson.com
Lars-Erik Jonsson
Ericsson Erisoft AB
Box 920
SE-971 28 Luleà
Sweden
Phone: +46 920 20 21 07
Email: lars-erik.jonsson@ericsson.com
Greg Sherwood
PacketVideo Corporation
4820 Eastgate Mall
San Diego, CA 92121
USA
Email: sherwood@packetvideo.com
Thomas Zeng
PacketVideo Corporation
4820 Eastgate Mall
San Diego, CA 92121
USA
Email: zeng@packetvideo.com
Adam H. Li [Page 17]