Internet Info 1994 March/Internet Info CD-ROM (Walnut Creek) (March 1994).iso/inet/internet-drafts/draft-ietf-avt-rtp-04.txt

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Internet Engineering Task Force                     Audio-Video Transport WG
INTERNET-DRAFT                                      H. Schulzrinne/S. Casner
draft-ietf-avt-rtp-04.txt                                           AT&T/ISI
                                                            October 20, 1993
                                                          Expires:  12/31/93

            RTP: A Transport Protocol for Real-Time Applications


Status of this Memo


This document is an Internet Draft.  Internet Drafts are working documents
of the Internet Engineering Task Force (IETF), its Areas, and its Working
Groups.   Note that other groups may also distribute working documents as
Internet Drafts.

Internet Drafts are draft documents valid for a maximum of six months.
Internet Drafts may be updated, replaced, or obsoleted by other documents
at any time.   It is not appropriate to use Internet Drafts as reference
material or to cite them other than as a ``working draft'' or ``work in
progress.''

Please check the I-D abstract listing contained in each Internet Draft
directory to learn the current status of this or any other Internet Draft.

Distribution of this document is unlimited.


                                  Abstract

     This memorandum describes the real-time transport protocol, RTP.
    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) designed to provide minimal
    control and identification functionality particularly in multicast
    networks.   Within multicast associations, sites can also direct
    control messages to individual sites.  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 bridges.


This specification is a product of the Audio/Video Transport working group
within the Internet Engineering Task Force.   Comments are solicited and
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should be addressed to the working group's mailing list at [e-mail]
and/or the authors.


Contents


1 Introduction                                                            3

2 RTP Use Scenarios                                                       5

  2.1 Simple Multicast Audio Conference. . . . . . . . . . . . . . . . . 5

  2.2 Bridges. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6

  2.3 Translators. . . . . . . . . . . . . . . . . . . . . . . . . . . . 6

  2.4 Security . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7

3 Definitions                                                             8

4 Byte Order, Alignment, and Reserved Values                             10

5 RTP Data Transfer Protocol                                             11

  5.1 RTP Fixed Header Fields. . . . . . . . . . . . . . . . . . . . . . 11

  5.2 The RTP Options. . . . . . . . . . . . . . . . . . . . . . . . . . 13

    5.2.1CSRC: Content source identifiers. . . . . . . . . . . . . . . . 13

    5.2.2SSRC: Synchronization source identifier . . . . . . . . . . . . 14

    5.2.3BOS: Beginning of synchronization unit. . . . . . . . . . . . . 15

  5.3 APP: Application-specific option . . . . . . . . . . . . . . . . . 15

  5.4 Reverse-Path Option. . . . . . . . . . . . . . . . . . . . . . . . 16

    5.4.1SDST: Synchronization destination identifier. . . . . . . . . . 17

  5.5 Security Options . . . . . . . . . . . . . . . . . . . . . . . . . 18

    5.5.1ENC: Encryption . . . . . . . . . . . . . . . . . . . . . . . . 21

    5.5.2MIC: Messsage integrity check . . . . . . . . . . . . . . . . . 22

    5.5.3MICA: Message integrity check, asymmetric encryption. . . . . . 22


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    5.5.4MICK: Message integrity check, keyed. . . . . . . . . . . . . . 23

    5.5.5MICS: Message integrity check, symmetric-key encrypted. . . . . 24

6 RTP Control Protocol --- RTCP                                          25

  6.1 FMT: Format description. . . . . . . . . . . . . . . . . . . . . . 25

  6.2 SDES: Source descriptor. . . . . . . . . . . . . . . . . . . . . . 26

  6.3 BYE: Goodbye . . . . . . . . . . . . . . . . . . . . . . . . . . . 30

  6.4 QOS: Quality of service measurement. . . . . . . . . . . . . . . . 30

7 Security Considerations                                                32

8 RTP over Network and Transport Protocols                               33

  8.1 Defaults . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33

  8.2 ST-II. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33

9 RTP Profiles                                                           34

A Implementation Notes                                                   35

  A.1 Timestamp Recovery . . . . . . . . . . . . . . . . . . . . . . . . 36

  A.2 Detecting the Beginning of a Synchronization Unit. . . . . . . . . 37

  A.3 Demultiplexing and Locating the Synchronization Source . . . . . . 38

  A.4 Parsing RTP Options. . . . . . . . . . . . . . . . . . . . . . . . 38

  A.5 Determining the Expected Number of RTP Packets . . . . . . . . . . 39

B Addresses of Authors                                                   40


1 Introduction


This  memorandum  specifies  the  real-time  transport  protocol  (RTP),
which  provides  end-to-end  delivery  services  for  data  with  real-time
characteristics, for example, interactive audio and video.   RTP itself
does not provide any mechanism to ensure timely delivery or provide other
quality-of-service guarantees, but relies on lower-layer services to do so.


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It does not guarantee delivery or prevent out-of-order delivery, nor does
it assume that the underlying network is reliable and delivers packets in
sequence.   The sequence numbers included in RTP allow the end system to
reconstruct the sender's packet sequence, but sequence numbers might also
be used to determine the proper location of a packet, for example in video
decoding, without necessarily decoding packets in sequence.  RTP is designed
to run on top of a variety of network and transport protocols, for example,
IP, ST-II or UDP.(1) RTP transfers data in a single direction, possibly to
multiple destinations if supported by the underlying network.  A mechanism
for sending control data in the opposite direction, reversing the path
traversed by regular data, is provided.

While RTP is primarily designed to satisfy the needs of multi-participant
multimedia conferences, it is not limited to that particular application.
Storage of continuous data,  interactive distributed simulation,  active
badge,  and  control  and  measurement  applications  may  also  find  RTP
applicable.   Profiles are used to instantiate certain header fields and
options for particular sets of applications (see Section 9).   A profile
for audio and video data may be found in the companion Internet draft
draft-ietf-avt-profile.

This document defines a packet format shared by two protocols:


  o the real-time transport protocol (RTP), for exchanging data that has
    real-time properties.    The RTP header consists of a fixed-length
    portion plus optional control fields;

  o the RTP control protocol (RTCP), for conveying information about the
    participants in an on-going session.   RTCP consists of additional
    header options that may be ignored without affecting the ability
    to receive data correctly.   RTCP is used for "loosely controlled"
    sessions, i.e., where there is no explicit membership control and
    set-up.    Its functionality may be subsumed by a session control
    protocol, which is beyond the scope of this document.


A discussion of real-time services and algorithms for their implementation
and background on some of the RTP design decisions can be found in the
current version of the companion Internet draft draft-ietf-avt-issues.

The current Internet does not support the widespread use of real-time
services.     High-bandwidth  services  using  RTP,  such  as  video,  can
potentially seriously degrade other network services.  Thus, implementors
should take appropriate precautions to limit accidental bandwidth usage.
Application  documentation  should  clearly  outline  the  limitations  and
possible operational impact of high-bandwidth real-time services on the
------------------------------
 1. For most applications, RTP offers insufficient demultiplexing to run
directly on IP.


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Internet and other network services.


2 RTP Use Scenarios


The following sections describe some aspects of the use of RTP. The examples
were chosen to illustrate the basic operation of applications using RTP,
not to limit what RTP may be used for.    In these examples, RTP is
carried on top of IP and UDP, and follows the conventions established by
the profile for audio and video specified in the companion Internet draft
draft-ietf-avt-profile.


2.1 Simple Multicast Audio Conference


A working group of the IETF meets to discuss the latest protocol draft,
using the IP multicast services of the Internet for voice communications.
Through some allocation mechanism,  the working group chair obtains a
multicast group address; all participants use the destination UDP port
specified by the profile.  The multicast address and port are distributed,
say, by electronic mail, to all intended participants.  The mechanisms for
discovering available multicast addresses and distributing the information
to participants are beyond the scope of RTP.

The audio conferencing application used by each conference participant sends
audio data in small chunks of, say, 20 ms duration.  Each chunk of audio
data is preceded by an RTP header; RTP header and data are in turn contained
in a UDP packet.  The Internet, like other packet networks, occasionally
loses and reorders packets and delays them by variable amounts of time.  To
cope with these impairments, the RTP header contains timing information and
a sequence number that allow the receivers to reconstruct the timing seen
by the source, so that, in our case, a chunk of audio is delivered to the
speaker every 20 ms.  The sequence number can also be used by the receiver
to estimate how many packets are being lost.  Each RTP packet also indicates
what type of audio encoding (such as PCM, ADPCM or GSM) is being used,
so that senders can change the encoding during a conference, for example,
to accommodate a new participant that is connected through a low-bandwidth
link.

Since members of the working group join and leave during the conference, it
is useful to know who is participating at any moment.  For that purpose,
each instance of the audio application in the conference periodically
multicasts the name, email address and other information of its user.  Such
control information is carried as RTCP SDES options within RTP messages,
with or without audio data (see Section 6.2).   These periodic messages
also provide some indication as to whether the network connection is still
functioning.   A site sends the RTCP BYE (Section 6.3) option when it
leaves a conference.  The RTCP QOS (Section 6.4) option indicates how well


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the current speaker is being received and may be used to control adaptive
encodings.


2.2 Bridges


So far, we have assumed that all sites want to receive audio data in the
same format.   However, this may not always be appropriate.   Consider
the case where participants in one area are connected through a low-speed
link to the majority of the conference participants, who enjoy high-speed
network access.   Instead of forcing everyone to use a lower-bandwidth,
reduced-quality audio encoding, a bridge is placed near the low-bandwidth
area.  This bridge resynchronizes incoming audio packets to reconstruct the
constant 20 ms spacing generated by the sender, mixes these reconstructed
audio streams, translates the audio encoding to a lower-bandwidth one and
forwards the lower-bandwidth packet stream to the low-bandwidth sites.

After the mixing, the identity of the high-speed site that is speaking can
no longer be determined from the network origin of the packet.  Therefore,
the bridge inserts a CSRC option (Section 5.2.1) into the packet containing
a list of short, locally unique site identifiers to indicate which site(s)
contributed to that mixed packet.  An example of this is shown for bridge B1
in Fig. 1.  As name and location information is received by the bridge in
SDES options from the high-speed sites, that information is passed on to the
receivers along with a mapping to the CSRC identifiers.  This also works
if an RTP packet is mixed through several bridges, with the CSRC value
being mapped into a new locally unique value at each bridge.  For example,
in Fig. 1 bridge B3 maps CSRC value 3 for packets coming from B2 into CSRC
value 1 for packets going to T2.


2.3 Translators


Not all sites are directly accessible through IP multicast.   For these
sites, mixing may not necessary, but a translation of the underlying
transport protocol is.   RTP-level gateways that do not restore timing or
mix packets from different sources are called translators in this document.
Application-level firewalls, for example, will not let any IP packets pass.
Two translators are installed, one on either side of the firewall, the
outside one funneling all multicast packets received through the secure
connection to the translator inside the firewall.   The translator inside
the firewall sends them again as multicast packets to a multicast group
restricted to the site's internal network.   Other examples include the
connection of a group of hosts speaking only IP/UDP to a group of hosts that
understand only ST-II.

After RTP packets have passed through a translator, they all carry the
network source address of the translator, making it impossible for the
receiver to distinguish packets from different speakers based on network

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      [E1]                                    [E6]
       |                                       |
 E1:17 |                                 E6:15 |
       |                                       |   E6:63/6
       V   B1:48 (1,2)         B1:28/1 (1,2)   V   B1:63/5 (1,2)
      (B1)-------------><T1>-----------------><T2>--------------->[E7]
       ^                 ^     E4:28/2         ^   E4:63/3
  E2:1 |           E4:47 |                     |   B3:63/4 (1,4)
       |                 |                     |
      [E2]              [E4]       B3:89 (1,4) |
                                               |            LEGEND:
[E3] --------->(B2)----------->(B3)------------|          [End system]
       E3:64        B2:12 (3)   ^                         (Bridge)
                                | E5:45                   <Translator>
                                |
                               [E5]          source: port/SSRC (CSRCs)
                                             ------------------------>

  Figure 1:  Sample RTP network with end systems, bridges and translators

source addresses.   Since each sending site has its own sequence number
space and slightly offset timestamp space, the receiver could not properly
mix the audio packets.   (For video, it could not properly separate them
into distinct displays.)   Instead of forcing all senders to include some
globally unique identifier in each packet, a translator inserts an SSRC
option (Section 5.2.2) with a short identifier for the source that is
locally unique to the translator.  This also works if an RTP packet has to
travel through several translators, with the SSRC value being mapped into a
new locally unique value at each translator.  An example is shown in Fig. 1,
where hosts T1 and T2 are translators.  The RTP packets from host E4 are
identified with SSRC value 2, while those coming from bridge B1 are labeled
with SSRC value 1.  Similarly, translator T2 has labeled packets from E6,
B1, E4 and B3 with SSRC values 6, 5, 3 and 4, respectively.


2.4 Security


Conference participants would often like to ensure that nobody else can
listen to their deliberations.  Encryption, indicated by the presence of the
ENC option (Section 5.5.1), provides that privacy.  The encryption method
and key can be changed during the conference by indexing into a table.  For
example, a meeting may go into executive session, protected by a different
encryption key accessible only to a subset of the meeting participants.

For authentication, a number of methods are provided, depending on needs and
computational capabilities.  All these message integrity check (MIC) options
(Sections 5.5.3 and following) compute cryptographic checksums, also known


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as message digests, over the RTP data.


3 Definitions


Payload is the data following the RTP fixed header and any RTP/RTCP options.
The payload format and interpretation are beyond the scope of this memo.
RTP packets without payload are valid.  Examples of payload include audio
samples and video data.

An RTP packet consists of the encapsulation specific to a particular
underlying protocol, the fixed RTP header, RTP and RTCP options, if any, and
the payload, if any.  A single packet of the underlying protocol may contain
several RTP packets if permitted by the encapsulation method.

A (protocol) port is the "abstraction that transport protocols use to
distinguish among multiple destinations within a given host computer.
TCP/IP protocols identify ports using small positive integers." [1] The
transport selectors (TSEL) used by the OSI transport layer are equivalent to
ports.

A transport address denotes the combination of network address, e.g., the
4-octet IP Version 4 address, and the transport protocol port, e.g., the UDP
port.  In OSI systems, the transport address is called transport service
access point or TSAP. The destination transport address may be a unicast or
multicast address.

A content source is the actual source of the data carried in an RTP
packet, for example, the application that originally generated some audio
data.  Data from one or more content sources may be combined into a single
RTP packet by a bridge, which becomes the synchronization source (see next
paragraph).  Content source identifiers carried in CSRC options identify the
logical source of the data, for example, to highlight the current speaker in
an audio conference; they have no effect on the delivery or playout timing
of the data itself.  In Fig. 1, E1 and E2 are the content sources of the
data received by E7 from bridge B1, while B1 is the synchronization source.

A synchronization source may be a single content source, or the combination
of one or more content sources, produced by a bridge, with its own timing.
Each synchronization source has its own sequence number space.  The audio
coming from a single microphone and the video from a camera are examples
of synchronization sources.  The receiver groups packets by synchronization
source for playback.   Typically a single synchronization source emits a
single medium (e.g., audio or video).  A synchronization source may change
its data format, e.g., audio encoding, over time.  Synchronization sources
are identified by their transport address and the identifier carried in the
SSRC option.  If the SSRC option is absent, a value of zero is assumed for
that identifier.



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A transport source is the transport-level origin of the RTP packets as seen
by the receiving end system.  In Fig. 1, host T2, port 63 is the transport
source of all packets received by end system E7.

A  channel  comprises  all  synchronization  sources  sending  to  the  same
destination transport address using the same RTP channel ID.

An end system generates the content to be sent in RTP packets and consumes
the content of received RTP. An end system can act as one or more
synchronization sources.   (Most end systems are expected to be a single
synchronization source.)  When a packet is transmitted from an end system,
the end system is the content source, synchronization source, and transport
source at that point.

An (RTP-level) bridge receives RTP packets from one or more sources,
combines them in some manner and then forwards a new RTP packet.   A
bridge may change the data format.  Since the timing among multiple input
sources will not generally be synchronized, the bridge will make timing
adjustments among the streams and generate its own timing for the combined
stream.  Therefore, when a packet is processed through a bridge, the bridge
becomes the synchronization source as well as the transport source, but the
originating end system remains the content source for that data.  As the
bridge combines packets from multiple content sources into a single outgoing
packet, each of the contributing content sources is noted by the insertion
of an identifier into the CSRC option in the outgoing packet.  Audio bridges
and media converters are examples of bridges.  In Fig. 1, end systems E1
and E2 use the services of bridge B1.   B1 inserts CSRC identifiers for
E1 and E2 when they are active (e.g., talking in an audio conference).
The RTP-level bridges described in this document are unrelated to the data
link-layer bridges found in local area networks.  If there is possibility
for confusion, the term `RTP-level bridge' should be used.  The name bridge
follows common telecommunication industry usage.

An (RTP-level) translator forwards RTP packets, but does not alter their
sequence numbers or timestamps.   Examples of its use include encoding
conversion without mixing or retiming, conversion from multicast to unicast,
and application-level filters in firewalls.    A translator is neither
a synchronization nor a content source, but does become the transport
source for packets which flow through it.    The properties of bridges
and translators are summarized in Table 1.   Checkmarks in parentheses
designate possible, but unlikely actions.  The RTP options are explained in
Section 5.2, the RTCP options in Section 6.

A synchronization unit consists of one or more packets that are emitted
contiguously by the sender.   The most common synchronization units are
talkspurts for voice and frames for video transmission.   During playout
synchronization, the receiver must reconstruct exactly the time difference
between packets within a synchronization unit (in the case of video, all the
packets of a frame are typically given the same timestamp so there is no
time difference).   The time difference between synchronization units may
be changed by the receiver to compensate for clock drift or to adjust to

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                                    end sys. bridge translator
           mix sources                --       x        --
           change encoding            --       x        x
           encrypt                     x       x       (x)
           sign for authentication     x       x        --
           alter content               x       x        x
           insert CSRC (RTP)          --       x        --
           insert SSRC (RTP)          (x)     (x)       x
           insert SDST (RTP)           x       x        --
           insert SDES (RTCP)          x       x        --


      Table 1:  The properties of end systems, bridges and translators

changing network delay jitter.  For example, if audio packets are generated
at fixed intervals during talkspurts, the receiver tries to play back
packets with exactly the same spacing.  However, if, for example, a silence
period between synchronization units (talkspurts) lasts 600 ms, the receiver
may adjust it to, say, 500 ms without this being noticed by the listener.

Non-RTP mechanisms refers to other protocols and mechanisms that may be
needed to provide a useable service.    In particular,  for multimedia
conferences,  a conference control application may distribute multicast
addresses  and  keys  for  encryption  and  authentication,  negotiate  the
encryption algorithm to be used,  and determine the mapping from the
RTP  format  field  to  the  actual  data  format  used.      For  simple
applications,  electronic  mail  or  a  conference  database  may  also  be
used.  The specification of such mechanisms is outside the scope of this
memorandum.


4 Byte Order, Alignment, and Reserved Values


All integer fields are carried in network byte order, that is, most
significant byte (octet) first.   This byte order is commonly known as
big-endian.  The transmission order is described in detail in [2], Appendix
A. Unless otherwise noted, numeric constants are in decimal (base 10).
Numeric constants prefixed by `0x' are in hexadecimal.

Fields within the fixed header and within options are aligned to the natural
length of the field, i.e., 16-bit words are aligned on even addresses,
32-bit long words are aligned at addresses divisible by four, etc.  Octets
designated as padding have the value zero.  Fields designated as "reserved"
or R are set aside for future use; they should be set to zero by senders and
ignored by receivers.

Textual information is encoded accorded to the UTF-2 encoding of the ISO


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standard 10646 (Annex F) [3,4].  US-ASCII is a subset of this encoding and
requires no additional encoding.  The presence of multi-octet encodings is
indicated by setting the most significant bit to a value of one.  An octet
with a binary value of zero may be used as a string terminator for padding
purposes.  However, strings are not required to be zero terminated.


5 RTP Data Transfer Protocol


5.1 RTP Fixed Header Fields


The RTP header has the following format:


 0                   1                   2                   3
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|Ver| ChannelID |P|S|  format   |       sequence number         |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|     timestamp (seconds)       |     timestamp (fraction)      |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| options ...                                                   |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+


The first eight octets are present in every RTP packet and have the
following meaning:


protocol version: 2 bits
    Identifies the protocol version.  The version number of the protocol
    defined in this memo is one (1).

channel ID: 6 bits
    The channel identifier field forms part of the tuple identifying a
    channel (see definition in Section 3) to provide an additional level of
    multiplexing at the RTP layer.   The channel ID field is convenient
    if several different channels are to receive the same treatment by
    the underlying layers or if a profile allows for the concatenation of
    several RTP packets on different channels into a single packet of the
    underlying protocol layer (see Section 8.1).

option present bit (P): 1 bit
    This flag has a value of one (1) if the fixed RTP header is followed by
    one or more options and a value of zero (0) otherwise.

end-of-synchronization-unit (S): 1 bit
    This flag has a value of one in the last packet of a synchronization


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    unit, a value of zero otherwise.  As shown in Appendix A, the beginning
    of a synchronization unit can be readily established from this flag.(2)

format: 6 bits
    The format field forms an index into a table defined through the
    RTCP  FMT  option  or  non-RTP  mechanisms  (see  Section  3).     The
    mapping establishes the format of the RTP payload and determines its
    interpretation by the application.   Formats defined through the FMT
    option must be kept in a separate mapping table per sender as there
    can be no guarantee that all senders will use the same table.  If no
    mapping has been defined through these mechanisms, a standard mapping
    is specified by the profile in use by the application in question.  An
    initial set of default mappings for audio and video is specified in
    the companion profile document RFC TBD, and may be extended in future
    editions of the Assigned Numbers RFC.

sequence number: 16 bits
    The sequence number counts RTP packets.  The sequence number increments
    by one for each packet sent.   The sequence number may be used by
    the receiver to detect packet loss, to restore packet sequence and to
    identify packets to the application.

timestamp: 32 bits
    The timestamp reflects the wall clock time when the RTP packet was
    generated.  Several consecutive RTP packets may have equal timestamps
    if they are generated at once.  The timestamp consists of the middle 32
    bits of a 64-bit NTP timestamp, as defined in RFC 1305 [5].  That is,
    it counts time since 0 hours UTC, January 1, 1900, with a resolution
    of 65536 ticks per second.    (UTC is Coordinated Universal Time,
    approximately equal to the historical Greenwich Mean Time.)  The RTP
    timestamp wraps around approximately every 18 hours.

    The timestamp of the first packet within a synchronization unit is
    expected to closely reflect the actual sampling instant,  measured
    by the local system clock.    If possible, the local system clock
    should be controlled by a time synchronization protocol such as NTP.
    However,  it is allowable to operate without synchronized time on
    those systems where it is not available, unless a profile or session
    protocol requires otherwise.   It is not necessary to reference the
    local system clock to obtain the timestamp for the beginning of
    every synchronization unit, but the local clock should be referenced
    frequently enough so that clock drift between the synchronized system
    clock and the sampling clock can be compensated for gradually.  Within
    one synchronization unit, it may be appropriate to compute timestamps
    based on the logical timing relationships between the packets.   For
    audio samples, for example, it is more accurate to maintain the time
------------------------------
 2. If this flag were to signal the beginning of a synchronization unit
instead, the end of a synchronization unit could not be established in real
time.


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    within a synchronization unit in samples, incrementing by the number
    of samples per packet, and then converting to an RTP timestamp (see
    Appendix A.1).


5.2 The RTP Options


The RTP fixed packet header may be followed by options and then the
payload.   Each option consists of the F (final) bit, the option type
designation, a one-octet length field denoting the total number of 32-bit
words comprising the option (including F bit, type and length), followed by
any option-specific data.  The last option before the payload has the F bit
set to one; for all other options this bit has a value of zero.

An application may discard options with types unknown to it.  The option
type number range is divided into four regions.   Types 0 through 31 are
general RTP options, their syntax and semantics independent of the format
or any profile.  Types 32 through 63 are RTCP options, again independent
of format and profile.  Types 64 through 95 are specific to a particular
profile, i.e., valid for a range of formats.   Types 96 through 126 are
specific to a format as defined by the four-character name registered
with the Internet Assigned Numbers Authority (IANA); these options may be
described in a profile or format specification.  Format-specific options are
parsed according to the format selected by the format field in the fixed
RTP header, as shown in Appendix A.4.  Types 0 through 126 are reserved and
registered with IANA. Type 127 is defined in Section 5.3 of this document;
it allows application-specific extensions not registered with IANA.

Unless otherwise noted, each option may appear only once per packet.  Each
packet may contain any number of options.  Options may appear in any order,
unless specifically restricted by the option description.  In particular,
the position of some security options has significance.


5.2.1 CSRC: Content source identifiers


 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F|  CSRC = 0   |    length     | content source identifiers   ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The  CSRC  option,  inserted  only  by  bridges,  lists  all  sources  that
contributed to the packet.  For example, for audio packets, all sources that
were mixed together to create a packet are enumerated, allowing correct
talker indication at the receiver.  The CSRC option may contain one or more
16-bit content source identifiers.  The identifier values must be unique for
all content sources received from a particular synchronization source on a
particular channel; the value of binary zero is reserved and may not be

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used.  If the number of content sources is even, the two octets needed to
pad the list to a multiple of four octets are set to zero.  There should
be no more than one CSRC option within a packet.   If no CSRC option is
present, the content source identifier is assumed to have a value of zero.
CSRC options are not modified by translators.   If a bridge receives a
packet containing a CSRC option from another bridge located upstream, the
identifier values in that CSRC option must be translated into new, locally
unique values.

A conformant RTP implementation does not have to be able to generate or
interpret the CSRC option.



5.2.2 SSRC: Synchronization source identifier


 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F|  SSRC = 1   |  length = 1   |          identifier           |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The SSRC option may be inserted by translators, end systems and bridges.
It is typically used only by translators, but it may be used by an
end system application to distinguish several sources sent with the same
transport source address.  If packets from multiple synchronization sources
will be transmitted with the same transport source address (e.g., the same
IP address and UDP port), an SSRC option must be inserted in each packet
with a distinct identifier for the synchronization source.   Conversely,
synchronization sources that are distinguishable by their transport address
do not require the use of SSRC options.  The SSRC value zero is reserved and
would not normally be transmitted; if received, the SSRC option should be
treated as if not present.  When no SSRC option is present, the transport
source address is assumed to indicate the synchronization source.   There
must be no more than one SSRC option per packet; thus, a translator must
remap the SSRC identifier of an incoming packet into a new, locally unique
SSRC identifier.   The SSRC option can be viewed as an extension of the
source port number in protocols like UDP, ST-II or TCP.

An RTP receiver must support the SSRC option.   RTP senders only need to
support this option if they intend to send more than one source to the same
channel using the same source port.  For example, a translator could use
multiple source ports rather than insert SSRC options, but this is likely to
be less convenient.







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5.2.3 BOS: Beginning of synchronization unit


 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F|   BOS = 3   |   length = 1  |        sequence number        |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The sequence number contained within this option is that of the first
packet within the current synchronization unit.  The BOS option allows the
receiver to compute the offset of a packet with respect to the beginning
of the synchronization unit, even if the last packet of the previous
synchronization unit was lost.  It is expected that many applications will
be able to tolerate such a loss, and so will not use the BOS option but rely
on the S bit.  Those applications which do require the BOS option may use a
profile that specifies it is always included.



5.3 APP: Application-specific option


 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F|  APP = 127  |    length     |            subtype            |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                          name (ASCII)                         |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                   application-dependent data                 ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


subtype: 2 octets
    May be used as a subtype to allow a set of APP options to be defined
    under one unique name, or for any application-dependendent data.

name: 4 octets
    A name chosen by the person defining the set of APP options to
    be unique with respect to other APP options this application might
    receive.  The application creator might choose to use the application
    name, and then coordinate the allocation of subtype values to others
    who want to define new options for the application.   Alternatively,
    it is recommended that others choose a name based on the entity they
    represent, then coordinate the use of the name within that entity.
    The name is interpreted as a sequence of four ASCII characters, with
    uppercase and lowercase characters treated as distinct.

application-dependent data: variable length
    Application-dependent data may or may not appear in an APP option.  It

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    is interpreted by the application and not RTP itself.


The APP option is intended for experimental use as new applications and new
features are developed, without requiring option type value registration.
APP options with unrecognized names should be ignored.   After testing
and if wider use is justified, it is recommended that each APP option
be redefined without the subtype and name fields and registered with the
Internet Assigned Numbers Authority using an option type in the RTP, RTCP,
profile-specific, or format-specific range as appropriate.



5.4 Reverse-Path Option


With two-party (unicast) communications, having a receiver of data relay
back control information to the sender is straightforward.  Similarly, for
multicast communications, control information can easily be sent to all
members of the group.   It may, however, be desirable to send a unicast
message to a single member of a multicast group, for example to send a
reception quality report.  For such purposes, RTP includes a mechanism for
sending so-called reverse RTP packets.  The format of reverse RTP packets is
exactly the same as for regular RTP packets and they can make use of any
option defined in this memorandum, except SSRC, as appropriate.  The support
for and semantics of particular options are to be specified by a profile or
an individual application.  Additional options may be defined as prescribed
in Section 5.2 as needed for a particular profile, format or application.

Reverse RTP packets travel through the same translators as forward RTP
packets.  When RTP is carried in a protocol that provides transport-level
addressing (ports), a site may distinguish reverse RTP packets from forward
RTP packets by their arrival port.  Reverse RTP packets arrive on the same
port that the site uses as a source port for forward (data) RTP packets.
Therefore, that port should be assigned uniquely; in particular, it should
be different than the destination port used with the multicast address,
and if the application is participating in several multicast groups, a
distinct source port should be used to send to each group.   If RTP is
carried directly within IP or some other network-layer protocol that does
not include port numbers, the reverse RTP packet must include an SDST option
(defined next), and the presence of the SDST option signals that the packet
is a reverse RTP packet.

A receiver of reverse RTP packets cannot rely on sequence numbers being
consecutive, as a sender is allowed to use the same sequence number space
while communicating through this reverse path with several sites.   In
particular, a receiver of reverse RTP packets cannot tell by the sequence
numbers whether it has received all reverse RTP packets sent to it.   As
a consequence, it is expected that reverse RTP packets would carry only
options and no payload.  The sequence number space of reverse RTP packets
has to be completely separate from that used for RTP packets sent to the

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multicast group.  If the same sequence number space were used, the members
of the multicast group not receiving reverse RTP packets would detect a
gap in their received sequence number space.   The sender of reverse RTP
packets should ensure that sequence numbers are unique, modulo wrap-around,
so that they can, if necessary, be used for matching request and response.
(Currently, no such request-response mechanism has been defined.)   As a
hypothetical example, consider defining a request to pan the remote video
camera.  After completing the request, the receiver of the request would
send a generic acknowledgement containing the sequence number of the request
back to the requestor as an option (not as the packet sequence number in the
fixed header).

The timestamp should reflect the approximate sending time of the packet.
The channel ID must be the same as that used in the corresponding forward
RTP packets.

If many receivers send a reverse RTP packet in response to a stimulus in the
data stream, for example a request for retransmission of a particular data
frame, the simultaneous delivery of a large number of packets back to the
data source can cause congestion for both the network and the destination
(this is known as an "ack implosion").  Thus reverse RTP packets should be
used with care, perhaps with mechanisms such as response rate limiting and
random delays to spread out the simultaneous delivery.


5.4.1 SDST: Synchronization destination identifier


 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F|  SDST = 2   |   length = 1  |           identifier          |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The SDST option is only inserted by RTP end systems and bridges if they want
to send a unicast packet to a particular site within the multicast group.
These are called reverse RTP packets.  Only reverse RTP packets may include
the SDST option, but not all reverse RTP packets require it, as explained
below.  A reverse RTP packet must not contain an SSRC option.

If a forward RTP packet carries SSRC identifier X when sent from A to
B, where A and B may be either two translators or an end system and a
translator, the unicast reverse RTP packet will carry an SDST option with
identifier X from B to A.

Consider the topology shown in Fig. 1.   Assume that RTP is carried over
a network or transport protocol that includes port numbers and that all
forward RTP packets are addressed to destination port 8000.  For the case
that B1 wants to send a reverse packet to E1, B1 simply sends to the source
address and port, that is, port 17 in this example.  E1 can tell by the
arrival on port 17 that the packet is a reverse packet rather than a regular

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(forward) packet.  No SDST option is required.

The mechanism is somewhat more complicated when translators intervene.  We
focus on end system E7.  E7 receives, say, video from a range of sources,
E1 through E6 as indicated by the arrows.   The transmission from T2 to
E7 could be either multicast or unicast.  Assume that E7 wants to send a
retransmission request, a request to pan the camera, etc., to end system E4
alone.   E7 may not be able to directly reach E4, as E4 may be using a
network protocol unknown to E7 or be located behind a firewall.  According
to the figure, video transmissions from E4 reach E7 through T2 with source
port 63 and SSRC identifier 3.  For the reverse message, E7 sends a message
to T2, with destination port 63 and SDST identifier 3.   T2 can look up
in its table that it sends forward data coming from T1 with that SSRC
identifier 3.  T2 also knows that those messages from T1 carry SSRC 2 and
arrive with source port 28.  Just like E7, T2 places the SSRC identifier, 2
in this case, into the SDST option and forwards the packet to T1 at port 28.
Finally, translator T1 consults its table to find that it labels packets
coming from E4, port 47 with SSRC value 2 and thus knows to forward the
reverse packet to E4, port 47.  T1 can either place value zero into the SDST
option or remove the option.  Note that E4 cannot directly determine that E7
sent the reverse packet, rather than, say, E6.   If that is important, a
global identifier as defined for the QOS option (Section 6.4) needs to be
included in some option in the reverse packet.

When reverse RTP packets are carried directly within IP or some other
network-layer protocol that does not include port numbers, the SDST option
is required to distinguish reverse RTP packets from forward RTP packets.  In
the case where no SSRC identifier needs to be placed in the SDST option, the
value zero should be inserted.

Only applications that need to send or receive reverse control RTP packets
need to implement the SDST option.



5.5 Security Options


The security options below offer message integrity, authentication and
confidentiality and the combination of the three.  Support for the security
options is not mandatory, but the encryption option (ENC) should at least be
recognized to avoid processing encrypted data.  The four message integrity
check options --- MIC, MICA, MICK and MICS --- are mutually exclusive, i.e.,
only one of them should be used in a single RTP packet.  Multiple options
are provided to satisfy varying security requirements and computational
capabilities.

A variety of security services may be provided with the encryption option,
one of the message integrity check options, or the combination of the two
options:


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confidentiality: Confidentiality means that only the intended receiver(s)
    can decode the received RTP packets;  for others,  the RTP packet
    contains no useful information.   Confidentiality of the content is
    achieved by encryption.  The presence of encryption and the encryption
    initialization vector is indicated by the ENC option.(3)

authentication and message integrity: These  two  security  services  are
    provided by the message integrity check options.   The receiver can
    ascertain that the claimed originator is indeed the originator of the
    data (authentication) and that the data within an RTP packet has not
    been altered after leaving the sender (message integrity).   Through
    examination of the timestamp and sequence number fields in the RTP
    header to verify that all the packets of a sequence are present and
    played in order, an implementation may also establish the integrity of
    that packet sequence.

    The  services  offered  by  MICA  and  MIC/MICK/MICS  differ:    With
    MIC/MICK/MICS, the receiver can only verify that the message originated
    within the group holding the secret key, rather than authenticate
    the sender of the message.   The MICA option, in combination with
    certificates(4),  affords true authentication of the sender.    The
    certificates for MICA must be distributed through means outside of RTP.

authentication, message integrity, and confidentiality: By  carrying  both
    the message integrity check and ENC option in RTP packets,  the
    authenticity, message integrity and confidentiality of the packet can
    be assured (subject to the limitations discussed in the previous
    paragraph).

    For this combination of security features when group authentication is
    sufficient, the combination ENC and MIC is recommended (instead of MICS
    or MICK), as it yields the lowest processing overhead.


All message integrity check options carry a message digest, which is a
cryptographic hash function that transforms a message of any length to a
fixed-length byte string, where the fixed-length string has the property
that it is computationally infeasible to generate another, different message
with the same digest.  The message digest is computed over the fixed header,
------------------------------
 3. For efficiency reasons,  this specification does not insist that
content encryption only be used in conjunction with message integrity and
authentication mechanisms, in which case there is no guarantee that the
encrypted data has not been replayed or rearranged.  This also means the
receiving program may not be able to readily determine whether the data has
been successfully decrypted, but in most cases, it will be obvious to the
person receiving the data if he or she does not possess the right encryption
key.
 4. For a description of certificates see, for example, RFC 1422 or [6].


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a portion of the message integrity check option itself (the first two octets
for MICA, the first four octets for MIC and MICS, or the whole option in
the case of MICK), and the remaining header options and payload that will
immediately follow the message integrity check option in the RTP packet.
The fixed header is protected to foil replay attacks and reassignment to a
different channel.

When both a message integrity check option and the ENC option are to be
included, the recommended ordering is that the message integrity check be
applied first as described in the previous paragraph.   Then the message
integrity check option and the remaining header options and payload that
follow it are encrypted using the shared secret key.   The ENC option is
prepended to the encrypted data; that is, the ENC option must be followed
immediately by the message integrity check option, without any other options
in between.  The receiver first decrypts the octets following the ENC option
and then authenticates the decrypted data using the message digest contained
in the message integrity check option.

For the MIC option in particular, this ordering must be used because the ENC
option is required to provide confidentiality of the message digest.  For
the other message integrity check options, this ordering allows explicit
detection of an encryption key mismatch.  However, both the decryption and
message integrity check functions must be performed before an invalid packet
can be detected, which increases the potential for a denial-of-service
attack.  For those applications where this is a concern, the ordering may be
reversed.

The message integrity check options and the ENC option must not cover the
SSRC or SDST option, i.e., SSRC or SDST must be inserted between the fixed
header and the ENC or message integrity check option; SSRC and SDST are
subject to change by translators that likely do not possess the necessary
descriptor table (see below) and encryption keys.   Trusted translators
that have the necessary keys and descriptor translation table may modify
the contents of the RTP packet, unless the MICA option is used (see MICA
description in Section 5.5.3).

All security options except MICA carry a one-octet descriptor field.  These
descriptors are indexes into two tables, one for the message integrity
check options and one for the ENC option, established by non-RTP means to
contain the digest algorithms (MD2, MD5, etc.), encryption algorithms (DES
variants) and encryption keys or shared secrets (for the MICK option) to be
used during a session.  The descriptor value may change during a session,
for example, to switch to a different encryption key.  The tables must be
established to be the same for all sources within the same channel; this
reduces per-site state information.

The descriptor value zero selects a set of default algorithms, namely,
MD5 for the message digest algorithm and DES in CBC mode for encryption
algorithm, so that basic security features may be implemented using simple
non-RTP mechanisms to communicate a single shared secret (key).    For
example, the key might be communicated by telephone or (private) email and

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entered manually.


5.5.1 ENC: Encryption


 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F|   ENC = 8   |   length = 3  |    reserved   |   descriptor  |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|      DES (CBC) initialization vector, bytes 0 through 3       |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|      DES (CBC) initialization vector, bytes 4 through 7       |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F|   ENC = 8   |   length = 1  |    reserved   |   descriptor  |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


Every encrypted RTP packet must contain this option in one of the two forms
shown.   All octets in the packet following this option are encrypted,
using the encryption key and symmetric encryption algorithm selected by the
descriptor field.  Note that the fixed header is specifically not encrypted
because some fields must be interpreted by translators that will not have
access to the key.  The descriptor value may change over time to accommodate
varying security requirements or limit the amount of ciphertext using the
same key.  For example, in a job interview conducted across a network, the
candidate and interviewers could share one key, with a second key set aside
for the interviewers only.  For symmetric keys, source-specific keys offer
no advantage.

The descriptor value zero is reserved for a default mode using the Data
Encryption Standard (DES) algorithm in CBC (cipher block chaining) mode,
as described in Section 1.1 of RFC 1423 [7].   The padding specified in
that section is to be used.   In the first form of the ENC option, the
8-octet initialization vector (IV) is carried unencrypted within the option,
but must be generated uniquely for each packet.    In the second form
(indicated by an option length of one), the ENC option does not contain
an initialization vector and instead the fixed RTP header is used as the
initialization vector.  (Using the fixed RTP header as the initialization
vector avoids regenerating the initialization vector for each packet and
incurs less header overhead; it is unique for a period of at least 18
hours.)   For details on the tradeoffs for CBC initialization vector use,
see [8].   Support for encryption is not required.   Implementations that
do not support encryption should recognize the ENC option so that they
can avoid processing encrypted messages and provide a meaningful failure
indication.  Implementations that support encryption should, at the minimum,
always support the DES algorithm in CBC mode.


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5.5.2 MIC: Messsage integrity check


 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F|   MIC = 9   |     length    |    reserved   |   descriptor  |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                 message digest (unencrypted)                 ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


The MIC option option is used only in combination with the ENC option
immediately preceding it to provide confidentiality, message integrity and
group membership authentication.   The message integrity check uses the
digest algorithm selected by the descriptor field.  The value zero implies
the use of the MD5 message digest.   Note that the MIC option is not
separately encrypted.



5.5.3 MICA: Message integrity check, asymmetric encryption


 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F|  MICA = 10  |    length     |         message digest       ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                    (asymmetrically encrypted)                ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The message digest is asymmetrically encrypted using the sender's private
key according to the algorithm defined for Privacy Enhanced Mail, described
in Section 4.2.1 of RFC 1423 (here "MIC" denotes the general term "message
integrity check", not the RTP option):


    As described in PKCS #1, all quantities input as data values to the
    RSAEncryption process shall be properly justified and padded to the
    length of the modulus prior to the encryption process.  In general,
    an RSAEncryption input value is formed by concatenating a leading
    NULL octet, a block type BT, a padding string PS, a NULL octet, and
    the data quantity D, that is, RSA input value = 0x00, BT, PS, 0x00,
    D. To prepare a MIC for RSAEncryption, the PKCS #1 "block type 01"
    encryption-block formatting scheme is employed.  The block type BT
    is a single octet containing the value 0x01 and the padding string
    PS is one or more octets (enough octets to make the length of the
    complete RSA input value equal to the length of the modulus) each
    containing the value 0xFF. The data quantity D is comprised of the
    MIC and the MIC algorithm identifier which are ASN.1 encoded.

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For the purposes of the MICA option, the encoding of data quantity D
may be considered as a fixed binary sequence identifying the message
integrity check algorithm, followed by the octets of the message digest.
Currently, only the use of the MD2 and MD5 algorithms is defined, as
described in RFC 1319 [9] (as corrected in Section 2.1 of RFC 1423) and RFC
1321 [10], respectively.  For MD2, the fixed binary sequence (shown here
in hexadecimal) is 0x30, 0x20, 0x30, 0x0C, 0x06, 0x08, 0x2A, 0x86, 0x48,
0x86, 0xF7, 0x0D, 0x02, 0x02, 0x05, 0x00, 0x04, 0x10, and for MD5 it is
0x30, 0x20, 0x30, 0x0C, 0x06, 0x08, 0x2A, 0x86, 0x48, 0x86, 0xF7, 0x0D,
0x02, 0x05, 0x05, 0x00, 0x04, 0x10.  The appropriate sequence is followed
immediately by the message digest, which is 16 octets long for both MD2 and
MD5.  For clarification of the octet ordering of the message digest, see RFC
1423, Sections 2.1 and 2.2.

As an example, for an RSA encryption modulus length of 512 bits or 64
octets, the RSA input value would be:


           BT  <---- PS --->      <------------ D ------------->
     0x00 0x01 0xFF ... 0xFF 0x00 0x30 ... 0x10 [message digest]
                (29 octets)        (18 octets)     (16 octets)

The length of the encrypted message digest will be equal to the modulus
of the RSA encryption used, rounded to the next integral octet count.
Contrary to what is specified in RFC 1423 for Privacy Enhanced Mail,
the asymmetrically encrypted message digest is carried in binary, not
represented in the printable encoding of RFC 1421, Section 4.3.2.4.  The
encrypted message digest is inserted into the MICA option immediately
following the length octet, and is padded at the end to make the MICA
option a multiple of four octets long.  The value of the padding is left
unspecified.  The number of non-padding bits within the signature is known
to the receiver as being equal to the key length.

The modulus and public key are conveyed to the receivers by non-RTP means.
After the message digest is decrypted, the message integrity check algorithm
is identified through the octets prepended to the actual 16-octet digest.

Asymmetric keys are used since symmetric keys would not allow authentication
of the individual source in the multicast case.  A translator is not allowed
to modify the parts of an RTP packet covered by the MICA option as the
receiver would have no way of establishing the identity of the translator
and thus could not verify the integrity of the RTP packet.



5.5.4 MICK: Message integrity check, keyed


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

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|F|  MICK = 11  |    length     |   reserved    |   descriptor  |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|           message digest (including shared secret)           ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

This message integrity check option does not require encryption,  but
includes a shared secret in the computation of the message digest.   The
shared secret is equivalent to the key used for the MICS and ENC options,
but is 16 octets long, padded if needed with binary zeroes.   The shared
secret is first placed into the MICK option where the message will later go,
then the digest is computed over the fixed RTP header, the whole MICK option
including the shared secret, and the remaining header options and payload
that will immediately follow the MICK option in the RTP packet.  The shared
secret in the MICK option is then replaced by the computed 16-octet message
digest for transmission.

The receiver stores the message digest contained in the MICK option,
replaces it with the shared secret key and computes the message digest in
the same manner as the sender.   If the RTP packet has not been tampered
with and has originated with one of the holders of the shared secret, the
computed message digest will agree with the digest found on reception in the
MICK option.(5)

The digest algorithm and shared secret are selected by the descriptor field.
The value zero implies the use of the MD5 message digest and a single shared
secret.



5.5.5 MICS: Message integrity check, symmetric-key encrypted


 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F|  MICS = 12  |    length     |   reserved    |   descriptor  |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|           message digest (symmetrically encrypted)           ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

This message integrity check encrypts the message digest using the DES
algorithm in ECB mode as described in RFC 1423, Section 3.1.  The digest
------------------------------
 5. This message integrity check follows the practice of SNMP Version 2,
as described in RFC 1446, Section 1.5.1.   Using the secret key in the
computation of the message digest instead of encrypting the digest avoids
the use of an encryption algorithm when only integrity and authentication
are desired.  However, the security of this approach has not been as well
established as the authentication based on encrypting message digests used
in the MICS, MIC and MICA options.


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algorithm and symmetric key are selected by the descriptor field.  The value
zero implies the use of the MD5 message digest and a single key.



6 RTP Control Protocol --- RTCP


The RTP control protocol (RTCP) conveys minimal control and advisory
information during a session.  It provides support for "loosely controlled"
sessions,  i.e.,  where participants enter and leave without membership
control and parameter negotiation.  The services provided by RTCP augment
RTP, but an end system does not have to implement RTCP features to
participate in sessions.   There is one exception to this rule:   if an
application sends FMT options, the receiver has to decode these in order
to properly interpret the RTP payload.   RTCP does not aim to provide
the services of a session control protocol and does not provide some of
the services desirable for two-party conversations.  If a session control
protocol is in use, the services of RTCP should not be required.  (As of the
writing of this document, a session or conference control protocol has not
been specified within the Internet.)

RTCP options share the same structure and numbering space as RTP options,
which are described in Section 5.    Unless otherwise noted,  control
information is carried periodically as options within RTP packets, with
or without payload.   RTCP packets are sent to all members of a session,
typically using multicast.   These packets are part of the same sequence
number space as RTP packets not containing RTCP options.  The period should
be varied randomly to avoid synchronization of all sources and its mean
should increase with the number of participants in the session to limit the
growth of the overall network and host interrupt load.  The length of the
period determines, for example, how long a receiver joining a session has
to wait until it can identify the source.  A receiver may remove from its
list of active sites a site that it has not been heard from for a given
time-out period; the time-out period may depend on the number of sites or
the observed average interarrival time of RTCP messages.   Note that not
every periodic message has to contain all RTCP options; for example, the
EMAIL part within the SDES option might only be sent every few messages.
RTCP options should also be sent when information carried in RTCP options
changes, but the generation of RTCP options should be rate-limited.


6.1 FMT: Format description


 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F|  FMT = 32   |    length     |R|R|  format   |    reserved   |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


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|                          format name                          |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                    format-dependent data                     ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


format: 6 bits
    The format field corresponds to the index value from the format field
    in the RTP fixed header, with values ranging from 0 to 63.

format name: 4 octets
    The format name describes the format in an unambiguous way and is
    registered with the Internet Assigned Numbers Authority.   The format
    name is interpreted as a sequence of four ASCII characters, with
    uppercase and lowercase characters treated as distinct.  Format names
    beginning with the letter 'X' are reserved for experimental use and not
    subject to registration.

format-dependent data: variable length
    Format-dependent data may or may not appear in a FMT option.  It is
    interpreted by the application and not RTP itself.


A FMT mapping changes the interpretation of a given format value carried in
the fixed RTP header starting at the packet containing the FMT option.  The
new interpretation applies only to packets from the same synchronization
source as the packet containing the FMT option.   If format mappings are
changed through the FMT option, the option should be sent periodically as
otherwise sites that did not receive the FMT option due to packet loss or
joining the session after the FMT option was sent will not know how to
interpret the particular format value.



6.2 SDES: Source descriptor


 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F|  SDES = 34  |    length     |       source identifier       |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| type=PORT (2) |   length = 1  |             port              |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| type=ADDR (1) |    length     |    reserved   | address type  |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                     network-layer address                    ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


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+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| type=PORT (2) |   length > 1  |    reserved   |    reserved   |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                             port                             ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| type=CNAME (4)|    length     | user and domain name         ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| type=EMAIL (5)|    length     | electronic mail address      ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| type=NAME (6) |    length     | common name of source        ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| type=LOC (8)  |    length     | geographic location of site  ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| type=TXT (16) |    length     | text describing source       ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| type=PRIV(255)|   length > 1  |            subtype            |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                          name (ASCII)                         |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                 application-defined content                  ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The SDES option is composed of the first four octets shown concatenated
with one or more of the subsequent items as described individually below.
SDES provides a mapping between a numeric source identifier and those items,
which describe a particular source.(6)   For those applications where the
size of a multi-item SDES option would be a concern, multiple SDES options
may be formed with subsets of the items to be sent in separate packets.

When an SDES option originates from a content source (the actual source
of the data), the identifier value is zero.   If the data flows through
a bridge, the bridge forwards the SDES information, but changes the SDES
source identifier to the value used in the CSRC option when identifying
------------------------------
 6. Several attributes were combined into one option so that the receiver
does not have to perform multiple mappings from identifiers to site data
structures.


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that content source.  The bridge may choose to store the SDES information
received from a content source and change then number of items sent together
or the rate at which SDES information is sent.  A bridge uses an identifier
value of zero within an SDES option to describe itself rather than the
content sources bridged by it, but if a bridge contributes local data to
outgoing packets, it should select another non-zero source identifier for
that traffic and send CSRC and SDES options for it as well.

Translators do not modify or insert SDES options.  The end system performs
the same mapping it uses to identify the content sources (that is, the
combination of transport source, SSRC identifier and the source identifier
within this SDES option) to identify a particular source.  SDES information
is specific to traffic from a source on a particular channel, unless a
profile or a higher-layer control protocol defines that the same SDES
describes traffic from that source on some set of channels.

Each item has a structure similar to that of RTP and RTCP options, that is,
a type field followed by a length field, measured in multiples of four
octets.  No final bit (see Section 5.2) is needed since the overall length
is known.   Item types 0 through 127 apply to all profiles, while types
128 through 254 are allocated to profile-specific items; both ranges are
reserved and registered with the Internet Assigned Numbers Authority (IANA).
Item type 255 (PRIV) is provided for private or experimental extensions not
registered with IANA. Items are independent of the format value.

All of the SDES items are optional and unrecognized items may be ignored;
however, if quality-of-service monitoring is to be used, receivers will
require the PORT and ADDR items from the SDES option in order to construct
the QOS option.   Only the TXT item is expected to change during the
duration of a session.  Text items are encoded according to the rules in
Section 4.  Items are padded with the binary value zero to the next multiple
of four octets.  Each item may appear only once unless otherwise noted.  A
description of the content of these items follows:


PORT/ADDR: The PORT item contains the source transport selector, such as
    the UDP source port number, and the ADDR item contains the network
    address of the source, for example, the IP version 4 address or an
    NSAP. Both are carried in binary form, not as "dotted decimal" or
    similar human-readable form.   Address types are identified by the
    Domain Name Service Resource Record (RR) type, as specified in the
    current edition of the Assigned Numbers RFC.

    There must be no more than one PORT item in an SDES option.  The PORT
    item should be followed immediately by an ADDR item.   Concatenated,
    these two items serve as a globally unique identifier for the source
    which is returned in the QOS option.  As far as RTP is concerned, this
    identifier is opaque, so it is unimportant which address is used for
    multi-homed hosts.  Applications may find the address or port useful
    for debugging or monitoring, but should not assume that the combination
    can be used to communicate with the source process because it may be on

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    the other side of a firewall or using a different transport protocol.
    If it is useful to the application, it is permissible for a source
    to include additional ADDR items after the first to convey additional
    addresses if the source is multi-homed, or if the source's address may
    be represented in multiple schemes, for example during the transition
    from IPv4 to IPng.(7)

    The figure shows the PORT item in two forms.  The first form shows the
    concatenated PORT and ADDR items as they would be used for the TCP
    and UDP protocols.  For an IPv4 address, the length of the ADDR item
    would be 2, and the address type would be 1.  The second form of the
    PORT item is indicated by a length field greater than one and is used
    when the transport selector (port number) is larger than two octets.
    Octets three and four of the item are reserved (zero) and the transport
    selector appears in words two and following of this item, in network
    byte order.

CNAME: Canonical user and host identifier, e.g.,


              "[e-mail]" or "sleepy.megacorp.com".


    The CNAME item must have the format "user@host" or "host", where
    "host" is the fully qualified domain name of the host from which the
    real-time data originates, formatted according to the rules specified
    in RFC 1034, RFC 1035 and Section 2.1 of RFC 1123.    The "host"
    form may be used if a user name is not available, for example on
    single-user systems.    The user name should be in a form that a
    program such as "finger" or "talk" could use, i.e., it typically is
    the login name rather than the real-life name.   Note that the host
    name is not necessarily identical to the electronic mail address of the
    participant.

EMAIL: User's electronic mail address, formatted according to RFC 822, for
    example,


                            "[e-mail]".


NAME: Common name describing the source, e.g., "John Doe, Bit Recycler,
    Megacorp".   This name may be in any form desired by the user.   For
    applications such as conferencing, this form of name may be the most
    desirable for display in participant lists, and therefore might be sent
    most frequently (profiles may establish such priorities).
------------------------------
 7. The ordering simplifies processing at the receiver, as the consecutive
octet string of PORT followed by the first ADDR can be stored as the
globally unique identifier.


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LOC: Geographic user location.   Depending on the application, different
    degrees of detail are appropriate for this item.    For conference
    applications,  a  string  like  "Murray  Hill,  New  Jersey"  may  be
    sufficient, while, for an active badge system, strings like "Room
    2A244, AT&T BL MH" might be appropriate.   The degree of detail is
    left to the implementation and/or user, but format and content may be
    prescribed by a profile.

TXT: Text describing the source, e.g., "out for lunch".

PRIV: Private, unregistered items.  The subtype and name fields are to be
    used in the same manner as in the APP option (Section 5.3).  The format
    and content of the octets following the name field are determined by
    the application defining the item.




6.3 BYE: Goodbye


 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F|   BYE = 35  | length = 1    |   content source identifier   |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The BYE option indicates that a particular session participant is no longer
active.   When a BYE option originates from a content source (the actual
source of the data), the identifier value is zero.    If the message
flows through a bridge, the bridge forwards the BYE message, but changes
the identifier to be the (non-zero) value used in the CSRC option when
identifying that content source.  If a bridge shuts down, it should first
send BYE options for all content sources it handles, followed by a BYE
option with an identifier value of zero.  An RTCP message may contain more
than one BYE option.  Multiple identifiers in a single BYE option are not
allowed, to avoid ambiguities between the special value of zero and any
necessary padding.



6.4 QOS: Quality of service measurement


 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F|  QOS = 36   |    length     |    reserved   |    reserved   |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                       packets expected                        |


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+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                       packets received                        |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|    minimum delay (seconds)    |    minimum delay (fraction)   |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|    maximum delay (seconds)    |    maximum delay (fraction)   |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|    average delay (seconds)    |    average delay (fraction)   |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| type=PORT (2) |    length     |   transport address          ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| type=ADDR (1) |    length     |    reserved   | address type  |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                     network-layer address                    ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


The QOS option conveys statistics on the reception of packets from a single
synchronization source on a single channel.  These statistics are the number
of packets received, the number of packets expected, the minimum delay, the
maximum delay and the average delay.  The expected number of packets may be
computed according to the algorithm in Section A.5.  The delay measures are
in units of 1/65536 of a second, that is, with the same resolution as the
timestamp in the fixed RTP header.

The  synchronization  source  to  which  these  statistics  correspond  is
identified by appending to the fixed-length part of the QOS option the
PORT and ADDR items, in that order, as received in an SDES option from
that source.   Together, the PORT and ADDR items form a globally unique
identifier (as described with the SDES option, Section 6.2).  If the source
has supplied more than one ADDR item, only the first one from the SDES (the
one immediately following the PORT item) is used.  If no SDES option, or
none with PORT and ADDR items, has been received from a particular source,
the QOS option cannot be sent unless the PORT and ADDR items are known by
some other mechanism.

The QOS option may be sent in either normal (forward) or reverse RTP
packets.    In the first case,  the channel to which these statistics
correspond is same as the channel on which the QOS option is sent; that is,
the channel is identified by the destination (multicast or unicast) address,
destination port and channel ID. If the QOS option is sent in a reverse RTP
packet, the channel is identified by the channel ID in the header and the
destination port number as the packet arrives at the synchronization source,
which will be the same port that the source uses to send data on that
channel, as described in Section 5.4.  Sending the QOS option by multicast
has the advantage that all participants in the session can compare their
reception to that of others, and allows participants to adjust the rate at
which QOS is sent based on the number of participants.

A single RTCP packet may contain several QOS options for different sources.
It is left to the implementor to decide how often to transmit QOS options

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and which sources are to be included.



7 Security Considerations


RTP suffers from the same security liabilities as the underlying protocols.
For example, an impostor can fake source or destination network addresses,
or change the header or payload.  For example, the SDES fields may be used
to impersonate another participant.  In addition, RTP may be sent via IP
multicast, which provides no direct means for a sender to know all the
receivers of the data sent and therefore no measure of privacy.  Rightly or
not, users may be more sensitive to privacy concerns with audio and video
communication than they have been with more traditional forms of network
communication [11].  Therefore, the use of security mechanisms with RTP is
important.

As a first step, RTP options make it easy for all participants in a session
to identify themselves; if deemed important for a particular application, it
is the responsibility of the application writer to make listening without
identification difficult.  It should be noted, however, that privacy of the
payload can generally be assured only by encryption.

The security options described in Section 5.5 can be used to implement
message integrity, authentication and confidentiality and the combination of
the three.  These security services might also be provided at the IP layer
as security mechanisms are developed for that layer.

The periodic transmission of SDES options from sources that are otherwise
idle may make it possible to detect denial-of-service attacks, as the
receiver can detect the absence of these expected messages.  The messages
that are received must be verified for integrity and authenticated before
being accepted for this purpose.

Unlike for other data, ciphertext-only attacks may be more difficult for
compressed audio and video sources.   Such data is very close to white
noise, making statistics-based ciphertext-only attacks difficult.   Even
without message integrity check options,  it may be difficult for an
attacker to detect automatically when he or she has found the secret
cryptographic key since the bit pattern after correct decryption may
not look significantly different from one decrypted with the wrong key.
However, the session information is more or less constant and predictable,
allowing known-plaintext attacks.    Chosen-plaintext attacks appear, in
general, to be difficult.

The integrity of the timestamp in the fixed RTP header can be protected by
the message integrity options.  If clocks are known to be synchronized, an
attacker only has a very limited time window of maybe a few seconds every 18
hours to replay recorded RTP without detection by the receiver.


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Key  distribution  and  certificates  are  outside  the  scope  of  this
document.


8 RTP over Network and Transport Protocols


This section describes issues specific to carrying RTP packets within
particular network and transport protocols.


8.1 Defaults


The following rules apply unless superseded by protocol-specific subsections
in this section.  The rules apply to both forward and reverse RTP packets.

RTP packets contain no length field or other delineation, therefore a
framing mechanism is needed if they are carried in underlying protocols that
provide the abstraction of a continuous bit stream rather than messages
(packets).  TCP is an example of such a protocol.  Framing is also needed
if the underlying protocol may contain padding so that the extent of the
RTP payload cannot be determined.   For these cases, each RTP packet is
prefixed by a 32-bit framing field containing the length of the RTP packet
measured in octets, not including the framing field itself.   If an RTP
packet traverses a path over a mixture of octet-stream and message-oriented
protocols, each RTP-level bridge between these protocols is responsible for
adding and removing the framing field.

A profile may specify that this framing method is to be used even when RTP
is carried in protocols that do provide framing in order to allow carrying
several RTP packets in one lower-layer protocol data unit, such as a UDP
packet.   Carrying several RTP packets in one network or transport packet
reduces header overhead and may simplify synchronization between different
streams.


8.2 ST-II


When used in conjunction with RTP, ST-II [12] service access ports (SAPs)
have a length of 16 bits.   The next protocol field (NextPCol, Section
4.2.2.10 in RFC 1190) is used to distinguish two encapsulations of RTP over
ST-II. The first uses NextPCol value TBD and directly places the RTP packet
into the ST-II data area.  If NextPCol value TBD is used, the RTP header is
preceded by a 32-bit header shown below.  The octet count determines the
number of octets in the RTP header and payload to be checksummed, allowing
the checksum to cover only the header if it is preferred to ignore errors in




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the data.  The 16-bit checksum uses the TCP and UDP checksum algorithm.


 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| count of octets to be checked |            checksum           |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|       RTP packet (fixed header, options and payload)         ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


9 RTP Profiles


RTP may be used for a variety of applications with somewhat differing
requirements.  The flexibility to adapt to those requirements is provided by
allowing multiple choices in the main protocol specification, then defining
a profile to select the appropriate choices for a particular class of
applications and environment.  A profile for audio and video applications
may be found in the companion Internet draft draft-ietf-avt-profile.

Within this specification, the following possible uses of a profile have
been identified, but this list is not meant to be exclusive:


  o Define a set of formats (e.g., media encodings) and a default mapping
    of those formats to format values.

  o Define new, application-class-specific options, or specify that an
    option, such as BOS, should be included in every packet.

  o Specify the support for and semantics of particular options to be used
    in Reverse Path messages.

  o Define new application-class-specific SDES items, or the data format,
    preferred use, or required use of particular SDES items.

  o Define when SDES applies to some grouping of channels rather than just
    a single channel.

  o Specify that globally synchronized time is required for operation of an
    application.

  o Specify  that  a  particular  underlying  network  or  transport  layer
    protocol will be used to carry RTP packets.

  o Specify that the RTP header is always to be prefixed by the framing
    field to allow carrying multiple RTP packets (perhaps for different
    media) in one lower-layer packet.

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  o Describe the application of the quality-of-service option.


It  is  not  expected  that  a  new  profile  will  be  required  for  every
application.    Within one application class,  it would be better to
extend an existing profile rather than make a new one.   For example,
additional options or formats can be defined and registered through IANA for
publication in the Assigned Numbers RFC as an alternative to publishing a
new profile specification.

It is recommended that a profile specify a default port number to be used
with that profile so that applications that support operation under multiple
profiles can use the port number to select the profile.


A Implementation Notes


We describe aspects of the receiver implementation in this section.  There
may be other implementation methods that are faster in particular operating
environments or have other advantages.  These implementation notes are for
informational purposes only.

The  following  definitions  are  used  for  all  examples;  the  structure
definitions are valid for 32-bit big-endian architectures only.  Bit fields
are assumed to be packed tightly, with no additional padding.


#include <sys/types.h>

typedef double CLOCK_t;

/* the definitions below are valid for 32-bit architectures and will
   have to be changed for 16- or 64-bit architectures */
typedef u_long  u_int32;
typedef u_short u_int16;

typedef enum {
  RTP_CSRC   = 0,
  RTP_SSRC   = 1,
  RTP_SDST   = 2,
  RTP_BOS    = 3,
  RTP_ENC    = 8,
  RTP_MIC    = 9,
  RTP_MICA   = 10,
  RTP_MICK   = 11,
  RTP_MICS   = 12,
  RTP_FMT    = 32,
  RTP_SDES   = 34,
  RTP_BYE    = 35,

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  RTP_QOS    = 36,
  RTP_MINFMT = 96,
  RTP_MAXFMT = 126,
  RTP_APP    = 127
} rtp_option_t;

typedef struct {
  unsigned int ver:2;      /* version number */
  unsigned int channel:6;  /* channel id */
  unsigned int p:1;        /* option present */
  unsigned int s:1;        /* sync bit */
  unsigned int format:6;   /* format of payload */
  u_int16 seq;             /* sequence number */
  u_int32 ts;              /* timestamp */
} rtp_hdr_t;

typedef union {
  struct {                 /* generic first 16 bits of options */
    unsigned int final:1;  /* final option */
    unsigned int type:7;   /* option type */
  } generic;
  struct {
    unsigned int final:1;  /* final option */
    unsigned int type:7;   /* option type */
    u_char length;         /* length, including type/length */
    u_int16 id[1];         /* content source identifier */
  } csrc;
  /* ... */
} rtp_t;


A.1 Timestamp Recovery


For some applications it is useful to have the receiver reconstruct the
sender's high-order bits of the NTP timestamp from the received 32-bit RTP
timestamp.  The following code uses double-precision floating point numbers
for whole numbers with a 48-bit range.  Other type definitions of CLOCK_t
may be appropriate for different operating environments,  e.g.,  64-bit
architectures or systems with slow floating point support.   The routine
applies to any clock frequency, not just the RTP value of 65,536 Hz, and any
clock starting point.  It will reconstruct the correct high-order bits as
long as the local clock now is within one half of the wrap-around time of
the 32-bit timestamp, e.g., approximately 9.1 hours for RTP timestamps.



#include <math.h>

#define MOD32bit 4294967296.
#define MAX31bit 0x7fffffff

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CLOCK_t clock_extend(ts, now)
u_int32 ts;   /* in: timestamp, low-order 32 bits */
CLOCK_t now;  /* in: current local time */
{
  u_int32 high, low;   /* high and low order bits of 48-bit clock */

  low  = fmod(now, MOD32bit);
  high = now / MOD32bit;

  if (low > ts) {
    if (low - ts > MAX31bit) high++;
  }
  else {
    if (ts - low > MAX31bit) high--;
  }
  return high * MOD32bit + ts;
} /* extend_timestamp */



Using the full timestamp internally has the advantage that the remainder of
the receiver code does not have to be concerned with modulo arithmetic.  The
current local time does not have to be derived directly from the system
clock for every packet.    For audio samples, for example, it is more
accurate to maintain the time within a synchronization unit in samples,
incrementing by the number of samples per packet, and then converting to an
RTP timestamp.  The following code implements the conversion from a 8 KHz
sample clock into an RTP timestamp.   This assumes that the sample clock
is also started at the RTP clock epoch (January 1, 1970).   If not, the
appropriate offset has to be added.


CLOCK_t t;       /* 8-kHz sample clock */
CLOCK_t RTP_ts;  /* RTP timestamp */

RTP_ts = floor(t * 8.192 + 0.5);


The whole seconds within NTP timestamps can be obtained by adding 2208988800
to the value of the standard Unix clock (generated, for example, by the
gettimeofday system call), which starts from the year 1970.  For the RTP
timestamp, only the least significant 16 bits of the seconds are used.


A.2 Detecting the Beginning of a Synchronization Unit


RTP packets contain a bit flag indicating the end of a synchronization unit.
The following code fragment determines, based on sequence numbers, if a

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packet is the beginning of a synchronization unit.   It assumes that the
packet header has been converted to host byte order.


static u_int32 seq_eos;
rtp_hdr_t *h;
static int flag;

if (h->s) {
  flag    = 1;
  seq_eos = h->seq;
}
/* handle wrap-around of sequence number */
else if (flag && (h->seq - seq_eos < 32768)) {
  flag = 0;
  /* code here to handle beginning of synchronization unit */
}


A.3 Demultiplexing and Locating the Synchronization Source


The combination of destination address, destination port and channel ID
determines the channel.  For each channel, the receiver maintains a list
of all sources, content and synchronization sources alike, in a table or
other suitable data structure.  Synchronization sources are stored with a
content source identifier of zero.  When an RTP packet arrives, the receiver
determines its network source address and port (from information returned
by the operating system), synchronization source identifier (SSRC option)
and content source identifier(s) (CSRC option).  To locate the table entry
containing timing information, mapping from content descriptor to actual
encoding, etc., the receiver sets the content source identifier to zero
and locates a table entry based on the tuple (transport source address,
synchronization source identifier, 0).

The receiver identifies the contributors to the packet (for example, the
speaker who is heard in the packet) through the list of content source
identifiers carried in the CSRC option.   To locate the table entry, it
matches on the triple (network address and port, synchronization source
identifier, content source identifier).


A.4 Parsing RTP Options


The following code segment walks through the RTP options,  preventing
infinite loops due to zero and invalid length fields.  Structure definitions
are valid for big-endian architectures only.




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u_int32 len;      /* length of RTP packet in bytes */
u_int32 *pt;      /* pointer */
rtp_hdr_t *h;     /* fixed header */
rtp_t *r;         /* options */

if (h->p) {
  pt = (u_int32 *)(h+1);
  do {
    r = (rtp_t *)pt;
    pt += r->generic.length;   /* point to end of option */

    /* invalid length field */
    if ((char *)pt - (char *)h > len || r->generic.length == 0) return -1;

    switch(r->generic.type) {
      case RTP_BYE:
        /* handle BYE option */
        break;
      case RTP_CSRC:
        /* handle CSRC option */
        break;

        /* ... */

      default:
                         if   (r->generic.type   >=   RTP_MINFMT   &&   r-
>generic.type <= RTP_MAXFMT) {
          /* call option handler particular to this format */
          (parse_format_options[h->format])(r);
        }
        else break;     /* ignore undefined options */
    }
  } while (!r->generic.final);
}


A.5 Determining the Expected Number of RTP Packets


The number of packets expected can be computed by the receiver by tracking
the first sequence number received (seq0),  the last sequence number
received, seq, and the number of complete sequence number cycles:


expected = cycles * 65536 + seq - seq0 + 1;


The cycle count is updated for each packet, where seq_prior is the sequence
number of the prior packet:


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unsigned long seq, seq_prior;

if (seq - seq_prior > 65536)
  cycle++;
else if (seq - seq_prior > 32768)
  cycle--;

seq_prior = seq;


Acknowledgments


This  memorandum  is  based  on  discussions  within  the  IETF  Audio/Video
Transport working group chaired by Stephen Casner.  The current protocol has
its origins in the Network Voice Protocol and the Packet Video Protocol
(Danny Cohen and Randy Cole) and the protocol implemented by the vat
application (Van Jacobson and Steve McCanne).   Stuart Stubblebine (ISI)
helped with the security aspects of RTP. Ron Frederick (Xerox PARC) provided
extensive editorial assistance.


B Addresses of Authors


Stephen Casner
USC/Information Sciences Institute
4676 Admiralty Way
Marina del Rey, CA 90292-6695
telephone:  [phone] (extension 153)
electronic mail:  [e-mail]


Henning Schulzrinne
AT&T Bell Laboratories
MH 2A244
600 Mountain Avenue
Murray Hill, NJ 07974-0636
telephone:  [phone]
facsimile:  [phone]
electronic mail:  [e-mail]


References


 [1] D. E. Comer, Internetworking with TCP/IP, vol. 1. Englewood Cliffs,
     New Jersey:  Prentice Hall, 1991.


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 [2] J. Postel, "Internet protocol," Network Working Group Request for
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 [3] International  Standards  Organization,  "ISO/IEC  DIS  10646-1:1993
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 [4] The Unicode Consortium, The Unicode Standard. New York, New York:
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 [5] D. L. Mills, "Network time protocol (version 3) -- specification,
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 [6] S.  Kent,  "Understanding  the  Internet  certification  system,"  in
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 [7] D. Balenson, "Privacy enhancement for internet electronic mail:  Part
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 [8] V. L. Voydock and S. T. Kent, "Security mechanisms in high-level
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 [9] J. Kaliski, Burton S., "The MD2 message-digest algorithm," Network
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[10] R. Rivest, "The MD5 message-digest algorithm," Network Working Group
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[11] S.  Stubblebine,  "Security  services  for  multimedia  conferencing,"
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[12] C. Topolcic, S. Casner, C. Lynn, Jr., P. Park, and K. Schroder,
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