RTP Payload Format for Tactical Secure Voice Cryptographic Interoperability Specification (TSVCIS) CodecVOCAL Technologies, Ltd.520 Lee Entrance, Suite 202BuffaloNY14228United States of America+1 716 688 4675victor.demjanenko@vocal.comVOCAL Technologies, Ltd.520 Lee Entrance, Suite 202BuffaloNY14228United States of America+1 716 688 4675john.punaro@vocal.comVOCAL Technologies, Ltd.520 Lee Entrance, Suite 202BuffaloNY14228United States of America+1 716 688 4675david.satterlee@vocal.com
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Payload Working GroupMELPMELPeTSVCISNRLVDRNaval Research LaboratoryNRLNATOTSVWGDepartment of DefenseDoDNSAMIL-STD
This document describes the RTP payload format for the Tactical
Secure Voice Cryptographic Interoperability Specification (TSVCIS)
speech coder. TSVCIS is a scalable narrowband voice coder supporting
varying encoder data rates and fallbacks. It is implemented as an
augmentation to the Mixed Excitation Linear Prediction Enhanced
(MELPe) speech coder by conveying additional speech coder parameters
to enhance voice quality. TSVCIS augmented speech data is
processed in conjunction with its temporally matched Mixed Excitation Linear
Prediction (MELP) 2400 speech data. The RTP packetization of TSVCIS and
MELPe speech coder data is described in detail.Status of This Memo
This is an Internet Standards Track document.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by
the Internet Engineering Steering Group (IESG). Further
information on Internet Standards is available in Section 2 of
RFC 7841.
Information about the current status of this document, any
errata, and how to provide feedback on it may be obtained at
.
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Table of Contents
. Introduction
. Conventions
. Abbreviations
. Background
. Payload Format
. MELPe Bitstream Definitions
. 2400 bps Bitstream Structure
. 1200 bps Bitstream Structure
. 600 bps Bitstream Structure
. Comfort Noise Bitstream Definition
. TSVCIS Bitstream Definition
. Multiple TSVCIS Frames in an RTP Packet
. Congestion Control Considerations
. Payload Format Parameters
. Media Type Definitions
. Mapping to SDP
. Declarative SDP Considerations
. Offer/Answer SDP Considerations
. Discontinuous Transmissions
. Packet Loss Concealment
. IANA Considerations
. Security Considerations
. References
. Normative References
. Informative References
Authors' Addresses
Introduction
This document describes how compressed Tactical Secure Voice
Cryptographic Interoperability Specification (TSVCIS) speech as
produced by the TSVCIS codec may be formatted for
use as an RTP payload. The TSVCIS speech coder (or TSVCIS speech-aware communications equipment on any intervening transport link) may
adjust to restricted bandwidth conditions by reducing the amount of
augmented speech data and relying on the underlying MELPe speech
coder for the most constrained bandwidth links.
Details are provided for packetizing the TSVCIS augmented speech data
along with MELPe 2400 bps speech parameters in an RTP packet. The
sender may send one or more codec data frames per packet, depending
on the application scenario or based on transport network conditions,
bandwidth restrictions, delay requirements, and packet loss
tolerance.Conventions
The key words "MUST", "MUST NOT",
"REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT",
"RECOMMENDED", "NOT RECOMMENDED",
"MAY", and "OPTIONAL" in this document are to be interpreted as
described in BCP 14
when, and only when, they appear in all capitals, as shown here.
Best current practices for writing an RTP payload format
specification were followed .AbbreviationsThe following abbreviations are used in this document.
Background
The MELP speech coder was developed by the US military as an upgrade
from the LPC-based CELP standard vocoder for low-bitrate
communications . ("LPC" stands for "Linear-Predictive Coding",
and "CELP" stands for "Code-Excited Linear Prediction".) MELP was
further enhanced and subsequently adopted by NATO as "MELPe" for use by
its members and Partnership for Peace countries for military and
other governmental communications as international NATO Standard
STANAG 4591 .
The Tactical Secure Voice Cryptographic Interoperability
Specification (TSVCIS) is a specification written by the Tactical
Secure Voice Working Group (TSVWG) to enable all modern tactical
secure voice devices to be interoperable across the US Department of
Defense . One of the most important aspects is that the
voice modes defined in TSVCIS are based on specific fixed rates of the Naval Research Lab's (NRL's) Variable Date Rate (VDR) Vocoder, which
uses the MELPe standard as its base . A complete TSVCIS
speech frame consists of MELPe speech parameters and corresponding
TSVCIS augmented speech data.
In addition to the augmented speech data, the TSVCIS specification
identifies which speech coder and framing bits are to be encrypted
and how they are protected by forward error correction (FEC)
techniques (using block codes). At the RTP transport layer, only the
speech coder-related bits need to be considered and are conveyed in
unencrypted form. In most IP-based network deployments, standard
link encryption methods (Secure Real-Time Transport Protocol (SRTP), VPNs, FIPS 140 link encryptors, or Type
1 Ethernet encryptors) would be used to secure the RTP speech
contents.
TSVCIS augmented speech data is derived from the signal processing
and data generated by the MELPe speech coder. For the
purposes of this specification, only the general parameter nature of
TSVCIS will be characterized. Depending on the bandwidth available
(and FEC requirements), a varying number of TSVCIS-specific speech
coder parameters need to be transported. These are first byte-packed
and then conveyed from encoder to decoder.
Byte packing of TSVCIS speech data into packed parameters is
processed as per the following example, where
Three-bit field:
Bits A, B, and C (A is MSB; C is LSB)
Five-bit field:
Bits D, E, F, G, and H (D is MSB; H is LSB)
MSB LSB
0 1 2 3 4 5 6 7
+------+------+------+------+------+------+------+------+
| H | G | F | E | D | C | B | A |
+------+------+------+------+------+------+------+------+
This packing method places the three-bit field "first" in the lowest
bits followed by the next five-bit field. Parameters may be split
between octets with the most significant bits in the earlier octet.
Any unfilled bits in the last octet MUST be filled with
zero.
In order to accommodate a varying amount of TSVCIS augmented speech
data, an octet count specifies the number of octets representing
the TSVCIS packed parameters. The encoding to do so is presented in
. TSVCIS specifically uses the NRL VDR in two
configurations with a fixed set of 15 and 35 packed octet
parameters in a standardized order .Payload Format
The TSVCIS codec augments the standard MELP 2400, 1200, and 600
bitrates and hence uses 22.5, 67.5, or 90 ms frames with a sampling
rate clock of 8 kHz, so the RTP timestamp MUST be in units of 1/8000
of a second.
The RTP payload for TSVCIS has the format shown in . No
additional header specific to this payload format is needed. This
format is intended for situations where the sender and the receiver
send one or more codec data frames per packet.
The RTP header of the packetized encoded TSVCIS speech has the
expected values as described in . The usage of the M bit
SHOULD be as specified in the applicable RTP profile -- for example,
specifies that if the sender does not
suppress silence (i.e., sends a frame on every frame interval), the
M bit will always be zero. When more than one codec data frame is
present in a single RTP packet, the timestamp specified is that of
the oldest data frame 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; if that
is not done, then a payload type in the dynamic range shall be chosen
by the sender.MELPe Bitstream Definitions
The TSVCIS speech coder includes all three MELPe coder rates used as
base speech parameters or as speech coders for bandwidth-restricted
links. RTP packetization of MELPe follows and is repeated
here for all three MELPe rates , with its recommendations now
regarded as requirements. The bits previously labeled as RSVA, RSVB,
and RSVC in SHOULD be filled with
rate code bits CODA, CODB,
and CODC, as shown in (compatible with Table 7 in ).
TSVCIS/MELPe Frame Bitrate Indicators and Frame Length
Coder Bitrate
CODA
CODB
CODC
Length
2400 bps
0
0
N/A
7
1200 bps
1
0
0
11
600 bps
0
1
N/A
7
Comfort Noise
1
0
1
2
TSVCIS Data
1
1
N/A
var.
The total number of bits used to describe one MELPe frame of 2400 bps
speech is 54, which fits in 7 octets (with two rate code bits). For
MELPe 1200 bps speech, the total number of bits used is 81, which
fits in 11 octets (with three rate code bits and four unused bits).
For MELPe 600 bps speech, the total number of bits used is 54, which
fits in 7 octets (with two rate code bits). The comfort noise frame
consists of 13 bits, which fits in 2 octets (with three rate code
bits). TSVCIS packed parameters will use the last code combination
in a trailing byte as discussed in .
It should be noted that CODB for MELPe 600 bps mode MAY deviate from
the value in when bit 55 is used as an alternating 1/0
end-to-end framing bit. Frame decoding would remain distinct as CODA
being zero on its own would indicate a 7-byte frame for either a 2400
or 600 bps rate, and the use of 600 bps speech coding could be deduced
from the RTP timestamp (and anticipated by the Session Description Protocol
(SDP) negotiations).2400 bps Bitstream Structure
The 2400 bps MELPe RTP payload is constructed as per . Note
that CODA MUST be filled with 0 and CODB SHOULD be filled with 0 as
per . CODB MAY contain an end-to-end framing bit if
required by the endpoints.1200 bps Bitstream Structure
The 1200 bps MELPe RTP payload is constructed as per . Note
that CODA, CODB, and CODC MUST be filled with 1, 0, and 0,
respectively, as per . RSV0 MUST be coded as 0.600 bps Bitstream Structure
The 600 bps MELPe RTP payload is constructed as per . Note
CODA MUST be filled with 0 and CODB SHOULD be filled with 1 as per
. CODB MAY contain an end-to-end framing bit if required
by the endpoints.Comfort Noise Bitstream Definition
The comfort noise MELPe RTP payload is constructed as per .
Note that CODA, CODB, and CODC MUST be filled with 1, 0, and 1,
respectively, as per .TSVCIS Bitstream Definition
The TSVCIS augmented speech data as packed parameters MUST be placed
immediately after a corresponding MELPe 2400 bps payload in the same
RTP packet. The packed parameters are counted in octets (TC). The
preferred placement SHOULD be used for TSVCIS payloads with TC less
than or equal to 77 octets; this is shown in . In the
preferred placement, a single trailing octet SHALL be appended to
include a two-bit rate code, CODA and CODB (both bits set to one),
and a six-bit modified count (MTC). The special modified count value
of all ones (representing an MTC value of 63) SHALL NOT be used for
this format as it is used as the indicator for the alternate packing
format shown next. In a standard implementation, the TSVCIS speech
coder uses a minimum of 15 octets for parameters in octet packed
form. The modified count (MTC) MUST be reduced by 15 from the full
octet count (TC). Computed MTC = TC-15. This accommodates a maximum
of 77 parameter octets (the maximum value of MTC is 62; 77 is the sum of
62+15).
In order to accommodate all other NRL VDR configurations, an
alternate parameter placement MUST use two trailing bytes as shown in
. The last trailing byte MUST be filled with a two-bit rate
code, CODA and CODB (both bits set to one), and its six-bit count
field MUST be filled with ones. The second to last trailing byte
MUST contain the parameter count (TC) in octets (a value from 1 and
255, inclusive). The value of zero SHALL be considered as reserved.Multiple TSVCIS Frames in an RTP Packet
A TSVCIS RTP packet payload consists of zero or more consecutive
TSVCIS coder frames (each consisting of MELPe 2400 and TSVCIS coder
data), with the oldest frame first, followed by zero or one MELPe
comfort noise frame. The presence of a comfort noise frame can be
determined by its rate code bits in its last octet.
The default packetization interval is one coder frame (22.5, 67.5, or
90 ms) according to the coder bitrate (2400, 1200, or 600 bps). For
some applications, a longer packetization interval is used to reduce
the packet rate.
A TSVCIS RTP packet without coder and comfort noise frames MAY be
used periodically by an endpoint to indicate connectivity by an
otherwise idle receiver.
TSVCIS coder frames in a single RTP packet MAY have varying TSVCIS
parameter octet counts. Its packed parameter octet count (length) is
indicated in the trailing byte(s). All MELPe frames in a single RTP
packet MUST be of the same coder bitrate. For all MELPe coder
frames, the coder rate bits in the trailing byte identify the
contents and length as per .
It is important to observe that senders have the following additional
restrictions:
Senders SHOULD NOT include more TSVCIS or MELPe frames in a single
RTP packet than will fit in the MTU of the RTP transport protocol.
Frames MUST NOT be split between RTP packets.
It is RECOMMENDED that the number of frames contained within an RTP packet
be consistent with the application. For example, in telephony and other
real-time applications where delay is important, the fewer frames per
packet, the lower the delay. However, for bandwidth-constrained links or
delay-insensitive streaming messaging applications, more than one frame per
packet or many frames per packet would be acceptable.
Information describing the number of frames contained in an RTP
packet is not transmitted as part of the RTP payload. The way to
determine the number of TSVCIS/MELPe frames is to identify each frame
type and length, thereby counting the total number of octets within
the RTP packet.Congestion Control Considerations
The target bitrate of TSVCIS can be adjusted at any point in time,
thus allowing congestion management. Furthermore, the amount of
encoded speech or audio data encoded in a single packet can be used
for congestion control, since the packet rate is inversely
proportional to the packet duration. A lower packet transmission
rate reduces the amount of header overhead but at the same time
increases latency and loss sensitivity, so it ought to be used
with care.
Since UDP does not provide congestion control, applications that use
RTP over UDP SHOULD implement their own congestion control above the
UDP layer and MAY also implement a transport circuit
breaker . Work in the RMCAT Working Group describes
the interactions and conceptual interfaces necessary between the
application components that relate to congestion control, including
the RTP layer, the higher-level media codec control layer, and the
lower-level transport interface, as well as components dedicated to
congestion control functions.Payload Format Parameters
This RTP payload format is identified using the TSVCIS media subtype,
which is registered in accordance with and per the
media type registration template from .Media Type Definitions
Type name:
audio
Subtype name:
TSVCIS
Required parameters:
Clock Rate (Hz): 8000
Optional parameters:
ptime:
the recommended length of time (in milliseconds)
represented by the media in a packet. It SHALL
use the nearest rounded-up ms integer packet duration. For
TSVCIS, this corresponds to the following values: 23, 45, 68,
90, 112, 135, 156, and 180. Larger values can be used as long
as they are properly rounded. See .
maxptime:
the maximum length of time (in milliseconds) that can be
encapsulated in a packet. It SHALL use the
nearest rounded-up ms integer packet duration. For TSVCIS, this
corresponds to the following values: 23, 45, 68, 90, 112, 135,
156, and 180. Larger values can be used as long as they are
properly rounded. See .
bitrate:
specifies the MELPe coder bitrates supported. Possible
values are a comma-separated list of rates from the following
set: 2400, 1200, 600. The modes are listed in order of
preference; the first is preferred. If "bitrate" is not
present, the fixed coder bitrate of 2400 MUST be
used.
tcmax:
specifies the TSVCIS maximum value for the TC supported or
desired, ranging from 1 to 255. If "tcmax" is not present, a
default value of 35 is used.
Channels:
1
Encoding considerations:
This media subtype is framed and binary; see .
Security considerations:
Please see of RFC 8817.
Interoperability considerations:
N/A
Published specification:
Applications that use this media type:
N/A
Fragment identifier considerations:
N/A
Additional information:
Deprecated alias names for this type:
N/A
Magic number(s):
N/A
File extension(s):
N/A
Macintosh file type code(s):
N/A
Person & email address to contact for further information:
<victor.demjanenko@vocal.com>
Intended usage:
COMMON
Restrictions on usage:
The media subtype depends on RTP
framing and hence is only defined for transfer via RTP . Transport within other framing protocols is not
defined at this time.
Author:
Change controller:
IETF; contact <avt@ietf.org>
Provisional registration? (standards tree only):
No
Mapping to SDP
The mapping of the above-defined payload format media subtype and its
parameters SHALL be done according to .
The information carried in the media type specification has a
specific mapping to fields in the Session Description Protocol (SDP)
, which is commonly used to describe RTP sessions. When SDP
is used to specify sessions employing the TSVCIS codec, the mapping
is as follows:
The media type ("audio") goes in SDP "m=" as the media name.
The media subtype (payload format name) goes in SDP "a=rtpmap" as
the encoding name.
The parameter "bitrate" goes in the SDP "a=fmtp" attribute by
copying it as a "bitrate=<value>" string.
The parameter "tcmax" goes in the SDP "a=fmtp" attribute by
copying it as a "tcmax=<value>" string.
The parameters "ptime" and "maxptime" go in the SDP "a=ptime" and
"a=maxptime" attributes, respectively.
When conveying information via SDP, the encoding name SHALL be
"TSVCIS" (the same as the media subtype).
An example of the media representation in SDP for describing TSVCIS
might be:
m=audio 49120 RTP/AVP 96
a=rtpmap:96 TSVCIS/8000
The optional media type parameter "bitrate", when present, MUST be
included in the "a=fmtp" attribute in the SDP, expressed as a media
type string in the form of a semicolon-separated list of
parameter=value pairs. The string "value" can be one or more of
2400, 1200, and 600, separated by commas (where each bitrate value
indicates the corresponding MELPe coder). An example of the media
representation in SDP for describing TSVCIS when all three coder
bitrates are supported might be:
m=audio 49120 RTP/AVP 96
a=rtpmap:96 TSVCIS/8000
a=fmtp:96 bitrate=2400,600,1200
The optional media type parameter "tcmax", when present, MUST be
included in the "a=fmtp" attribute in the SDP, expressed as a media
type string in the form of a semicolon-separated list of
parameter=value pairs. The string "value" is an integer number in
the range of 1 to 255 representing the maximum number of TSVCIS
parameter octets supported. An example of the media representation
in SDP for describing TSVCIS with a maximum of 101 octets supported
is as follows:
m=audio 49120 RTP/AVP 96
a=rtpmap:96 TSVCIS/8000
a=fmtp:96 tcmax=101
The parameter "ptime" cannot be used for the purpose of specifying
the TSVCIS operating mode due to the fact that, for certain values, it
will be impossible to distinguish which mode is about to be used
(e.g., when ptime=68, it would be impossible to distinguish whether the
packet is carrying one frame of 67.5 ms or three frames of 22.5 ms).
Note that the payload format (encoding) names are commonly shown in
upper case. Media subtypes are commonly shown in lower case. These
names are case insensitive in both places. Similarly, parameter
names are case insensitive in both the media subtype name and the
default mapping to the SDP a=fmtp attribute.Declarative SDP Considerations
For declarative media, the "bitrate" parameter specifies the possible
bitrates used by the sender. Multiple TSVCIS rtpmap values (such as
97, 98, and 99, as used below) MAY be used to convey TSVCIS-coded
voice at different bitrates. The receiver can then select an
appropriate TSVCIS codec by using 97, 98, or 99.
m=audio 49120 RTP/AVP 97 98 99
a=rtpmap:97 TSVCIS/8000
a=fmtp:97 bitrate=2400
a=rtpmap:98 TSVCIS/8000
a=fmtp:98 bitrate=1200
a=rtpmap:99 TSVCIS/8000
a=fmtp:99 bitrate=600
For declarative media, the "tcmax" parameter specifies the maximum
number of octets of TSVCIS packed parameters used by the sender or the
sender's communications channel.Offer/Answer SDP Considerations
In the Offer/Answer model , "bitrate" is a bidirectional
parameter. Both sides MUST use a common "bitrate" value or values.
The offer contains the bitrates supported by the offerer, listed in
its preferred order. The answerer MAY agree to any bitrate by
listing the bitrate first in the answerer response. Additionally,
the answerer MAY indicate any secondary bitrate or bitrates that it
supports. The initial bitrate used by both parties SHALL be the
first bitrate specified in the answerer response.
For example, if offerer bitrates are "2400,600" and answerer bitrates
are "600,2400", the initial bitrate is 600. If other bitrates are
provided by the answerer, any common bitrate between the offer and
answer MAY be used at any time in the future. Activation of these
other common bitrates is beyond the scope of this document.
The use of a lower bitrate is often important for a case such as when
one endpoint utilizes a bandwidth-constrained link (e.g., 1200 bps
radio link or slower), where only the lower coder bitrate will work.
In the Offer/Answer model , "tcmax" is a bidirectional
parameter. Both sides SHOULD use a common "tcmax" value. The offer
contains the tcmax supported by the offerer. The answerer MAY agree
to any tcmax equal to or less than this value by stating the desired
tcmax in the answerer response. The answerer alternatively MAY
identify its own tcmax and rely on TSVCIS ignoring any augmented data
it cannot use.Discontinuous Transmissions
A primary application of TSVCIS is for radio communications of voice
conversations, and discontinuous transmissions are normal. When
TSVCIS is used in an IP network, TSVCIS RTP packet transmissions may
cease and resume frequently. RTP synchronization source (SSRC)
sequence number gaps indicate lost packets to be filled by Packet
Loss Concealment (PLC), while abrupt loss of RTP packets indicates
intended discontinuous transmissions. Resumption of voice
transmission SHOULD be indicated by the RTP marker bit (M) set to 1.If a TSVCIS coder so desires, it may send a MELPe comfort noise frame as
per Appendix B of prior to ceasing transmission. A
receiver may optionally use comfort noise during its silence periods. No
SDP negotiations are required.
Packet Loss Concealment
TSVCIS packet loss concealment (PLC) uses the special properties and
coding for the pitch/voicing parameter of the MELPe 2400 bps coder.
The PLC erasure indication utilizes any of the errored encodings of a
non-voiced frame as identified in Table 1 of . For the sake of
simplicity, it is preferred that a code value of 3 for the
pitch/voicing parameter be used. Hence, set bits P0 and P1 to one
and bits P2, P3, P4, P5, and P6 to zero.
When using PLC in 1200 bps or 600 bps mode, the MELPe 2400 bps
decoder is called three or four times, respectively, to cover the
loss of a low bitrate MELPe frame.IANA Considerations
IANA has registered TSVCIS as specified in . The media type has been added to the IANA
registry for "RTP Payload Format Media Types"
().Security Considerations
RTP packets using the payload format defined in this specification
are subject to the security considerations discussed in the RTP
specification and in any applicable RTP profile such as
RTP/AVP , RTP/AVPF , RTP/SAVP , or
RTP/SAVPF . However, as discussed in , it is not
an RTP payload format's responsibility to discuss or mandate what
solutions are used to meet such basic security goals as
confidentiality, integrity, and source authenticity for RTP in
general. This responsibility lies with anyone using RTP in an
application. They can find guidance on available security mechanisms
and important considerations in . Applications SHOULD use
one or more appropriate strong security mechanisms. The rest of this
section discusses the security-impacting properties of the payload
format itself.
This RTP payload format and the TSVCIS decoder, to the best of our
knowledge, do not exhibit any significant non-uniformity in the
receiver-side computational complexity for packet processing and thus
are unlikely to pose a denial-of-service threat due to the receipt of
pathological data. Additionally, the RTP payload format does not
contain any active content.
Please see the security considerations discussed in
regarding Voice Activity Detect (VAD) and its effect on bitrates.ReferencesNormative ReferencesAnalog-to-Digital Conversion of Voice by 2,400 Bit/Second Mixed Excitation Linear Prediction (MELP)Department of DefenseThe 600 Bit/S, 1200 Bit/S and 2400 Bit/S NATO Interoperable Narrow Band Voice CoderNorth Atlantic Treaty Organization (NATO)Universal Vocoder Using Variable Data Rate VocodingKey words for use in RFCs to Indicate Requirement LevelsIn many standards track documents several words are used to signify the requirements in the specification. These words are often capitalized. This document defines these words as they should be interpreted in IETF documents. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.Guidelines for Writers of RTP Payload Format SpecificationsThis document provides general guidelines aimed at assisting the authors of RTP Payload Format specifications in deciding on good formats. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.An Offer/Answer Model with Session Description Protocol (SDP)This document defines a mechanism by which two entities can make use of the Session Description Protocol (SDP) to arrive at a common view of a multimedia session between them. In the model, one participant offers the other a description of the desired session from their perspective, and the other participant answers with the desired session from their perspective. This offer/answer model is most useful in unicast sessions where information from both participants is needed for the complete view of the session. The offer/answer model is used by protocols like the Session Initiation Protocol (SIP). [STANDARDS-TRACK]RTP: A Transport Protocol for Real-Time ApplicationsThis memorandum describes RTP, the real-time transport protocol. RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services. RTP does not address resource reservation and does not guarantee quality-of- service for real-time services. The data transport is augmented by a control protocol (RTCP) to allow monitoring of the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality. RTP and RTCP are designed to be independent of the underlying transport and network layers. The protocol supports the use of RTP-level translators and mixers. Most of the text in this memorandum is identical to RFC 1889 which it obsoletes. There are no changes in the packet formats on the wire, only changes to the rules and algorithms governing how the protocol is used. The biggest change is an enhancement to the scalable timer algorithm for calculating when to send RTCP packets in order to minimize transmission in excess of the intended rate when many participants join a session simultaneously. [STANDARDS-TRACK]RTP Profile for Audio and Video Conferences with Minimal ControlThis document describes a profile called "RTP/AVP" for the use of the real-time transport protocol (RTP), version 2, and the associated control protocol, RTCP, within audio and video multiparticipant conferences with minimal control. It provides interpretations of generic fields within the RTP specification suitable for audio and video conferences. In particular, this document defines a set of default mappings from payload type numbers to encodings. This document also describes how audio and video data may be carried within RTP. It defines a set of standard encodings and their names when used within RTP. The descriptions provide pointers to reference implementations and the detailed standards. This document is meant as an aid for implementors of audio, video and other real-time multimedia applications. This memorandum obsoletes RFC 1890. It is mostly backwards-compatible except for functions removed because two interoperable implementations were not found. The additions to RFC 1890 codify existing practice in the use of payload formats under this profile and include new payload formats defined since RFC 1890 was published. [STANDARDS-TRACK]The Secure Real-time Transport Protocol (SRTP)This document describes the Secure Real-time Transport Protocol (SRTP), a profile of the Real-time Transport Protocol (RTP), which can provide confidentiality, message authentication, and replay protection to the RTP traffic and to the control traffic for RTP, the Real-time Transport Control Protocol (RTCP). [STANDARDS-TRACK]SDP: Session Description ProtocolThis memo defines the Session Description Protocol (SDP). SDP is intended for describing multimedia sessions for the purposes of session announcement, session invitation, and other forms of multimedia session initiation. [STANDARDS-TRACK]Media Type Registration of RTP Payload FormatsThis document specifies the procedure to register RTP payload formats as audio, video, or other media subtype names. This is useful in a text-based format description or control protocol to identify the type of an RTP transmission. [STANDARDS-TRACK]Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)An RTP profile (SAVP) for secure real-time communications and another profile (AVPF) to provide timely feedback from the receivers to a sender are defined in RFC 3711 and RFC 4585, respectively. This memo specifies the combination of both profiles to enable secure RTP communications with feedback. [STANDARDS-TRACK]Guidelines for the Use of Variable Bit Rate Audio with Secure RTPThis memo discusses potential security issues that arise when using variable bit rate (VBR) audio with the secure RTP profile. Guidelines to mitigate these issues are suggested. [STANDARDS-TRACK]Media Type Specifications and Registration ProceduresThis document defines procedures for the specification and registration of media types for use in HTTP, MIME, and other Internet protocols. This memo documents an Internet Best Current Practice.Multimedia Congestion Control: Circuit Breakers for Unicast RTP SessionsThe Real-time Transport Protocol (RTP) is widely used in telephony, video conferencing, and telepresence applications. Such applications are often run on best-effort UDP/IP networks. If congestion control is not implemented in these applications, then network congestion can lead to uncontrolled packet loss and a resulting deterioration of the user's multimedia experience. The congestion control algorithm acts as a safety measure by stopping RTP flows from using excessive resources and protecting the network from overload. At the time of this writing, however, while there are several proprietary solutions, there is no standard algorithm for congestion control of interactive RTP flows.This document does not propose a congestion control algorithm. It instead defines a minimal set of RTP circuit breakers: conditions under which an RTP sender needs to stop transmitting media data to protect the network from excessive congestion. It is expected that, in the absence of long-lived excessive congestion, RTP applications running on best-effort IP networks will be able to operate without triggering these circuit breakers. To avoid triggering the RTP circuit breaker, any Standards Track congestion control algorithms defined for RTP will need to operate within the envelope set by these RTP circuit breaker algorithms.UDP Usage GuidelinesThe User Datagram Protocol (UDP) provides a minimal message-passing transport that has no inherent congestion control mechanisms. This document provides guidelines on the use of UDP for the designers of applications, tunnels, and other protocols that use UDP. Congestion control guidelines are a primary focus, but the document also provides guidance on other topics, including message sizes, reliability, checksums, middlebox traversal, the use of Explicit Congestion Notification (ECN), Differentiated Services Code Points (DSCPs), and ports.Because congestion control is critical to the stable operation of the Internet, applications and other protocols that choose to use UDP as an Internet transport must employ mechanisms to prevent congestion collapse and to establish some degree of fairness with concurrent traffic. They may also need to implement additional mechanisms, depending on how they use UDP.Some guidance is also applicable to the design of other protocols (e.g., protocols layered directly on IP or via IP-based tunnels), especially when these protocols do not themselves provide congestion control.This document obsoletes RFC 5405 and adds guidelines for multicast UDP usage.How to Write an RTP Payload FormatThis document contains information on how best to write an RTP payload format specification. It provides reading tips, design practices, and practical tips on how to produce an RTP payload format specification quickly and with good results. A template is also included with instructions.RTP Payload Format for the Mixed Excitation Linear Prediction Enhanced (MELPe) CodecThis document describes the RTP payload format for the Mixed Excitation Linear Prediction Enhanced (MELPe) speech coder. MELPe's three different speech encoding rates and sample frame sizes are supported. Comfort noise procedures and packet loss concealment are described in detail.Ambiguity of Uppercase vs Lowercase in RFC 2119 Key WordsRFC 2119 specifies common key words that may be used in protocol specifications. This document aims to reduce the ambiguity by clarifying that only UPPERCASE usage of the key words have the defined special meanings.SCIP Signaling PlanNational Security AgencySCIP-210Informative ReferencesExtended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)Real-time media streams that use RTP are, to some degree, resilient against packet losses. Receivers may use the base mechanisms of the Real-time Transport Control Protocol (RTCP) to report packet reception statistics and thus allow a sender to adapt its transmission behavior in the mid-term. This is the sole means for feedback and feedback-based error repair (besides a few codec-specific mechanisms). This document defines an extension to the Audio-visual Profile (AVP) that enables receivers to provide, statistically, more immediate feedback to the senders and thus allows for short-term adaptation and efficient feedback-based repair mechanisms to be implemented. This early feedback profile (AVPF) maintains the AVP bandwidth constraints for RTCP and preserves scalability to large groups. [STANDARDS-TRACK]Options for Securing RTP SessionsThe Real-time Transport Protocol (RTP) is used in a large number of different application domains and environments. This heterogeneity implies that different security mechanisms are needed to provide services such as confidentiality, integrity, and source authentication of RTP and RTP Control Protocol (RTCP) packets suitable for the various environments. The range of solutions makes it difficult for RTP-based application developers to pick the most suitable mechanism. This document provides an overview of a number of security solutions for RTP and gives guidance for developers on how to choose the appropriate security mechanism.Securing the RTP Framework: Why RTP Does Not Mandate a Single Media Security SolutionThis memo discusses the problem of securing real-time multimedia sessions. It also explains why the Real-time Transport Protocol (RTP) and the associated RTP Control Protocol (RTCP) do not mandate a single media security mechanism. This is relevant for designers and reviewers of future RTP extensions to ensure that appropriate security mechanisms are mandated and that any such mechanisms are specified in a manner that conforms with the RTP architecture.RTP Media Congestion Avoidance Techniques (rmcat) Working GroupIETFTactical Secure Voice Cryptographic Interoperability Specification (TSVCIS) Version 3.1National Security AgencyAuthors' AddressesVOCAL Technologies, Ltd.520 Lee Entrance, Suite 202BuffaloNY14228United States of America+1 716 688 4675victor.demjanenko@vocal.comVOCAL Technologies, Ltd.520 Lee Entrance, Suite 202BuffaloNY14228United States of America+1 716 688 4675john.punaro@vocal.comVOCAL Technologies, Ltd.520 Lee Entrance, Suite 202BuffaloNY14228United States of America+1 716 688 4675david.satterlee@vocal.com