Internet Draft Adam H. Li
draft-ietf-avt-evrc-06.txt UCLA
July 20, 2001 Editor
Expires: January 20, 2002
An RTP Payload Format for EVRC Speech
STATUS OF THIS MEMO
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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. An 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 .... 4
4.1. Packet Table of Content entries .............................. 4
4.2. The Codec Data Frame ......................................... 5
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5. Bundling Codec Data Frames in Type 1 Packets ................... 6
6. Interleaving Codec Data Frame in Type 1 Packets ................ 7
6.1. Finding Interleave Group Boundaries .......................... 8
6.2. Reconstructing Interleaved Speech ............................ 8
6.3. Receiving Invalid Interleaving Values ........................ 9
6.4. Additional Receiver Responsibilities ......................... 9
7. Handling Lost RTP Packets ...................................... 9
8. Implementation Issues ......................................... 10
8.1. Interleaving Length ......................................... 10
8.2. Signaling of Reduce Rate .................................... 10
9. IANA Considerations ........................................... 10
9.1 Storage Mode ................................................. 11
9.2 EVRC MIME Registration ....................................... 11
10. Mapping to SDP Parameters .................................... 12
11. Security Considerations ...................................... 12
12. Acknowledgements ............................................. 13
13. References ................................................... 13
14. AuthorsÆ Address ............................................. 13
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 bundled/interleaved 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 [3].
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 K bits/sec 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 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 and/or bundling to send one or more than
one codec 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 [2] |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| RR| LLL | NNN | |
+-+-+-+-+-+-+-+-+ one or more ToC entries +-------------+
| | |
+-------------------------------------------------+ |
| |
| one or more codec data frames |
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The RTP header has the expected values as described in [2]. The M bit
should be set as specified in the applicable RTP profile, for
example, RFC 1890. Note that RFC 1890 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.
The fields of the interleaving byte 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 7 inclusive.
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.
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Table of Content field (ToC) contains the indexes for the codec data
frame(s) in the packet. There is one entry for each codec data frame.
More than one codec data frame MAY be included in a single RTP packet
by a sender. The data frames may be included in one of the two
following manners: bundled or interleaved. Bundling of the codec data
frames is described in detail in Section 5, and interleaving in
Section 6.
3.2 Type 2 RTP/EVRC Packet Format
The Type 2 RTP/EVRC Packet Format is designed for maximum efficiency
in transmission of the EVRC codec data. Only one codec data frame is
sent with each RTP packet, and there is no ToC field prefix the codec
data. The EVRC codec rate of the data frame can be found out at the
receiver from the length of the codec frame, since there is only one
codec data frame in each RTP packet for this type.
The RTP header for Type 2 RTP/EVRC Packet Format is the same as
described in Section 3.1 for Type 1 RTP/EVRC Packet 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] |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| |
+ ONLY one codec data frames +-+-+-+-+-+-+-+-+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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.
The packets of the two types can be distinguished by checking the
payload type field in the RTP header. 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 Table
of Content (ToC) entry associated with it. The ToC entry indicates
whether interleaving is present, if rate reduction is desired, if
there are more entries following the current one, and the rate of the
corresponding codec frame. Type 2 packets do NOT have the ToC field,
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since 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
Indicate if there are more ToC entries following the current on
or the current one is the last in the ToC entry field. F = 1
indicates there are more ToC entries following. F = 0 indicates
that the current entry is the last one in ToC.
Reduce Rate (D): 1 bit
Setting the 'D' bit indicates that this packet is requesting a
reduced codec rate for the reverse direction. When the 'D' bit is
not set the packet 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 packet 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 are described in the table below and the
size of the associated packet is indicated 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)
Receipt of a ToC entry with a reserved value in Frame Type MUST
be considered invalid data. All values not listed in the above
table MUST be considered reserved.
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 follows:
The bits as numbered in the standard [1] from the lowest to the
highest 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
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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 as the others fit completely
into a whole number of octets).
Here is a detail of how a Rate 1 frame is converted into a CODEC data
frame:
The codec data frame for a Rate 1 frame is 22 byte long. Bits 1
through 171 from the standard Rate 1 frame are placed as indicated
with bits marked with "Z" being set to zero. The Rate 1/8 and 1/2
standard 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| | | | | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
5. Bundling 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 RTP packet by a sender. Bundling codec data
frames means multiple data frames are included consecutively in a
packets without interleaving. The bundling of codec data frames is
signaled by setting the LLL value in the Interleaving Byte to 0.
Senders MAY support bundling. All receivers MUST support bundling.
Receivers MAY signal the maximum number of codec data frames they can
handle in a single RTP packet.
Furthermore, senders have the following additional restrictions:
o MUST never bundle more codec data frames in a single RTP packet
than signaled by maxptime in Section 9.
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o SHOULD 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.
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 the ToC field of the RTP packet until the entry with F bit
set to 0 is reached.
6. Interleaving Codec Data Frames in Type 1 Packets
Senders MAY support interleaving. All receivers MUST support
interleaving. Receivers MAY signal the maximum number of codec data
frames they can handle in a single RTP packet. Interleaving of codec
data frames is signaled by setting the LLL value in the Interleaving
Byte to a value between 1 and 7 inclusive.
Given a time-ordered sequence of output frames from the EVRC 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.
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Additionally, senders have the following restrictions:
o Once beginning a session with a given maximum interleaving value
set by maxinterleave in Section 9, MUST NOT increase the
interleaving value exceeding the maximum interleaving the value
that is signaled.
o MAY change the interleaving value only between interleave groups.
6.1 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.
6.2 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 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
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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
6.3 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.
6.4 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 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.
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 interleaved speech or playing back non-interleaved
speech, erasure frames MUST be fed to the 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 bundling/interleaving 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 should be treated as containing B
erasure CODEC data frames where B is the bundling value for that
interleave group.
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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, the non-interleaved
RTP payload type is recommended.
When end-to-end delay is not a concern, an interleaving value (field
LLL) of 4 or 5 is recommended subject to MTU limitations.
The parameters maxptime and maxinterleaving 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 value, 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
signal MUST be discarded.
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.
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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., 0x2321455652430a.
The speech codec frames are stored in consecutive order with the TOC
entry byte prefix each codec frame data.
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 keep synchronization with the original media.
9.2 EVRC MIME Registration
Media Type Name: audio
Media Subtype Name: EVRC
Required Parameters:
ptype: It is the type of the RTP/EVRC packets. The valid
values are 1 or 2.
Optional parameters for RTP mode:
ptime: Defined as usual for RTP audio.
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 size. If not signaled, the default maxptime value is 200
milliseconds.
maxinterleave: Maximum number for interleaving value. The
interleaving values used in the entire session should not
exceed this maximum value. If not signaled, the maxinterleave
value is 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: The EVRC speech frames are
packed into consecutive compound EVRC payloads, see Section 5 and
Section 6 of RFC xxxx. The compound EVRC payloads must be stored
in sequential order. Furthermore, missing frames and non-received
frames during non-speech period must be encapsulated into a
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compound EVRC payload as blank frames or erasures. Each receiving
entity that accepts this MIME type must be able to decode all
EVRC coding modes.
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.
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 chapter 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; maxptime=4
11. 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 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
be overloaded. However, this encoding does not exhibit any
significant non-uniformity.
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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.
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] 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, RFC 2119, March 1997.
[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
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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
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
Adam H. Li [Page 14]
INTERNET-DRAFT An RTP Payload Format for EVRC Speech July 20, 2001
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
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
Room 2F-218
263 Shuman Drive
Naperville, IL 60137
USA
Phone: +1 630 979 7673
Email: tom.hiller@lucent.com
Peter J. McCann
Lucent Technologies
Room 2Z-305
263 Shuman Drive
Naperville, IL 60137
USA
Phone: +1 630 713 9359
Email: mccap@lucent.com
Michael D. Turner
Lucent Technologies
Room 2A-203
67 Whippany Rd
Whippany, NJ 07981
USA
Phone: +1 973 386 3579
Email: mdturner@lucent.com
Ajay Rajkumar
Lucent Technologies
Room 1A-235
Adam H. Li [Page 15]
INTERNET-DRAFT An RTP Payload Format for EVRC Speech July 20, 2001
67 Whippany Rd
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
Magnus Westerlund
Ericsson Research
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 Lule
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 16]
