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INTERNET-DRAFT Bernard Aboba
<draft-aboba-rtp-http-02.txt> Microsoft Corporation
Payload Format for HTTP Encoding in RTP
1. Status of this Memo
This document is an Internet-Draft. Internet-Drafts are working docu-
ments of the Internet Engineering Task Force (IETF), its areas, and
its working groups. Note that other groups may also distribute work-
ing documents as Internet-Drafts.
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The distribution of this memo is unlimited. It is filed as <draft-
aboba-rtp-http-02.txt>, and expires July 1, 1997. Please send com-
ments to the authors.
2. Abstract
This document specifies a payload format for use in encoding HTTP
within RTP. This payload format can be used for unreliable multicast-
ing of resources such as Web pages, stock tickers, etc. As with other
RTP applications, receiver feedback and group membership information
is provided via RTCP. This specification is not expected to be used
with unicast, since unicast applications will instead use HTTP over
TCP.
3. Introduction
3.1. Purpose
Considerable interest has recently arisen in the multicasting of
resources residing on HTTP servers. Many of these uses can be satis-
fied by unreliable multicast transport.
In this context, unreliable refers to applications where a repair
mechanism is not required. These are typically applications where the
material is of time value (stock tickers), so that it makes more sense
to wait for the resource to be re-multicast than to attempt to repair
it; applications in which an alternative means is available for
retrieving the resource (cache filling); applications in which error
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correction is performed at the datalink layer; or applications in
which a separate error correction stream is transmitted along with the
data, typically on a separate group.
In a cache filling application, there is a relatively small probabil-
ity that a particular missing resource will be hit, and so it is often
more costly to request a repair than to leave an incompletely received
resource in the cache, where it may never be requested. Cache hits
when they do occur, will typically be spread out over time, and there-
fore not synchronized to the original transmission. As a result, such
applications present no danger of a NAK implosion.
The encoding specified in this document is not appropriate for use in
applications where reliable transmission is required. Such applica-
tions present the possibility of a NAK implosion or congestive col-
lapse, and so must be carefully analyzed prior to deployment.
3.2. Requirements
Before discussing the proposed HTTP encoding in RTP, it is useful to
describe the requirements for unreliable multicast transmission of
resources:
Source differentiation
Resource demultiplexing
Receiver reporting
Sender reporting
Layered encoding
Ability to synchronize with other media
Low overhead
3.2.1. Source differentiation
Since mixing is not useful for transmission of resources, and allowing
multiple sources would make it difficult to maintain rate control, it
is likely that only one source will be sending to a group at one time.
Nevertheless, in the case where there is a handoff, it is necessary to
be able to differentiate sources, since packets from the two sources
may be intermingled.
3.2.2. Resource demultiplexing
For multicast resource transmission, it is not desirable to have each
resource transmitted with a unique source ID. Resources are typically
of small size, and therefore the overhead of obtaining a source ID and
setting up the transmission would be excessive. As a result, multiple
resources will typically be transmitted with a single source ID. Since
it is possible for packets from one resource to become intermingled
with another due to out of order delivery, it is necessary to be able
to demultiplex resources within a single source ID.
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3.2.3. Receiver reporting
Although the encoding described in this document is to be used only
for unreliable transmission, receiver feedback may still be desirable.
Such feedback can be used to estimate listenership, packet loss rates,
and receiver bandwidth availability.
Typically, receiver reporting information will be used both for engi-
neering purposes (diagnosis of transmission problems) as well as for
business purposes (listenership information). While receiver reports
could be useful in allowing senders to adjust transmission parameters,
typically it is more desirable to allow receiver-driven rate adapta-
tion via layered encoding.
By transmitting a resource on several groups, each starting transmis-
sion with a different offset, the receiver may adjust their reception
rate based on the available bandwidth. Typically, the group transmis-
sion rates will be tailored to commonly available bandwidths, i.e. 10
Kbps for 14.4 Kbps modems, 20 Kbps for 28.8 Kbps, 30 Kbps for single
channel ISDN, etc.
Sender-driven transmission rate adjustment appears to be useful in
only a limited number of circumstances. In cases where a small frac-
tion of listeners are experiencing problems, it is undesirable to
adjust the transmission rate; instead, the affected receivers should
adjust their rate by leaving the higher bandwidth groups. If this
does not work, they should stop listening to the transmission alto-
gether.
A circumstance in which sender-driven transmission rate adjustment
appears useful is in the case where the majority of listeners are only
subscribed to the lowest transmission rate group, yet appear to lack
the bandwidth to also join an error correction group appropriate to
their packet loss rate. In this case the sender should back off the
transmission rate on the lowest group to allow for successful recep-
tion of the error correction information.
As there are applications in which receiver feedback may not be feasi-
ble or desirable (satellite transmission), it must be possible to turn
off the receiver reporting mechanism if desired.
3.2.4. Sender reporting
Just as receivers may wish to provide feedback to senders, so may
senders wish to provide instructions to receivers. This may include
information about the particular resource being transmitted (such as
the resource length, related keywords, or URL), or information on the
status of the transmission itself, such as the highest offset trans-
mitted.
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3.2.5. Layered encoding
Layered multicasting appears to be essential for multicast transmis-
sion of resources, since it provides for receiver-driven rate adapta-
tion, as well as for transmission of error-correction information.
While layered encoding was originally proposed for use in audio/video
transmission, it can also be applied to data carousel style transmis-
sions. In this application, each of the layers transmits the same
resource, beginning at a different offset. Receivers with sufficient
bandwidth may then subscribe to multiple groups, and receive the
resource more quickly.
Layered encoding may also provide for error correction, by allowing
error correction information to be transmitted on separate groups.
Receivers may then subscribe to these groups according to their aver-
age packet loss rate; receivers experiencing high loss rates will typ-
ically join a higher bandwidth error correction group. In order to
allow for the additional bandwidth of an error correction group, the
sender transmission rate should be set appropriately.
3.2.6. Ability to synchronize with other media
While most uses of unreliable resource multicasting do not have real-
time requirements, this may not be true of all such applications. For
example, it may be desirable to synchronize display of a resource with
an audio/video stream, as in a presentation or lecture, and in such
an application it may be desirable to include an encoded clock.
3.2.7. Low Overhead
Since resource multicasting will typically use a small MTU size (i.e.
536 octets), it is important that a low overhead encapsulation be cho-
sen. In order to achieve this, the GET request must not be sent in
every packet. In addition, it may be desirable to support header com-
pression.
3.3. Why RTP?
Given that RTP was originally created for use in realtime applica-
tions, it is not entirely obvious that it is the appropriate protocol
to use for resource multicasting. However, given that this applica-
tion appears to require receiver and sender reporting, layered encod-
ing, and source and resource de-multiplexing, it is likely that an
alternative framing will end up more closely resembling RTP than a
simple UDP-based approach.
Where an alternative encoding would be most likely to differ would be
in its reporting mechanisms. For example while the basic header of
RMFP, described in [9], bears considerable resemblance to RTP, the
sender and receiver packet formats differ considerably from RTCP.
However, given the current state of knowledge, it is far from clear
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that the reporting needs of unreliable resource multicasting differ
substantially from those provided by RTCP.
Thus while it is not apparent that RTP is the ideal protocol for use
in this application, it appears to meet the application requirements.
In addition, the behavior of RTP is well understood, and the protocol
provides for functions such as RTP monitoring and header compression.
3.4. Overview
Multicasting of HTTP payloads is useful in a variety of applications,
and as a result, several approaches have been taken. One of these is
to put a single resource within a payload; another is to pack multiple
resources in a payload. If multiple resources are placed within a sin-
gle payload, this can be accomplished either via encapsulation or via
a packing method such as MHTML, defined in [5].
This document specifies encapsulation of a single resource per pay-
load. As defined in this specification, the HTTP payload consists of
a preamble header, a MIME-like header, and entity-body content. Infor-
mation on the resource being transmitted (such as the URI and the off-
set) is placed in the preamble header so as to avoid requiring
receivers to parse MIME-headers in the HTTP payload in order to deter-
mine what portions of a resource have been received. As a result, this
specification does not propose any new MIME headers, and any MIME
headers allowable in an HTTP GET response may be enclosed in the pay-
load.
This simplifies construction of unicast-multicast proxies, since the
MIME-like header in the payload can be identical to that returned in
the response to an HTTP-GET request. Proxies may therefore make a
request for a URL, stuff the response into a payload, and multicast
it. This is an efficient process since the proxy does not need to
parse the response or construct an MHTML package.
4. RTP header
Rather than defining a new profile, this specification assumes that
HTTP payloads defined in [8] will be transmitted using the RTP profile
defined in [3], that is the RTP profile for audio and video conferenc-
ing. Additional required parameters are accommodated via definition
of an HTTP payload format. As a result, there is no need for header
extensions, SDES private extensions, new sender or receiver report
fields, new RTCP packet types, or changes in the reporting interval
constants.
Nevertheless, some clarifications are useful in describing how HTTP
payload encoding is to be used with the profile defined in [3].
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4.1. Extension bit
Although this specification does not preclude use of header exten-
sions, it does not define any such extensions, and thus it is expected
that the extension bit will typically be cleared.
4.2. CSRC count
Since HTTP payloads do not require mixing, there is no need for a con-
tributing source field. As a result, the CSRC count field is set to
zero.
4.3. Marker Bit
For use with HTTP payloads, a zero marker bit is used to indicate the
last packet for a resource. The first packet is distinguished by a
zero offset, and as a result, does not need to be marked.
4.4. Payload types
A dynamic payload type in the range of 96-128 is used. The binding of
payload type and application is accomplished by non-RTP means, such as
use of the "m=" and "a=" fields of the session description:
m=data 32768 RTP/AVP 98
a=rtpmap: 98 MHTTP
4.5. Port numbers
Although there is no official policy on this, current practice dic-
tates usage of port ranges as follows:
[0, 16384] - lowest priority, unclassified
[16384, 32768] - highest priority, i.e. audio
[32768, 49152] - medium priority, i.e. whiteboard
[49152, 65536] - low priority, i.e. video
It is recommended that unreliable multicast HTTP use the medium prior-
ity port range.
4.6. SSRC
Unlike with A/V payloads, a sender may use a single synchronization
source for transmission of multiple HTTP payload streams. Thus a JPEG
file may be transmitted with the same SSRC as an HTML file. This does
not present a problem since the timestamp is used to uniquely identify
resource streams.
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4.7. Timestamp
Since a single synchronization source is used for transmission of mul-
tiple resources, an additional parameter is needed to identify packets
within the same resource. The URI was not felt to be sufficient for
this purpose, since a given resource may be multicast in data-carousel
style, changing with each transmission.
As a result, the RTP timestamp field is used for this purpose. For
use with HTTP payloads, the timestamp is constant for all packets that
form a single transmission of a resource, and represents the time at
which the sender began to transmit the resource.
Due to out of order delivery it is possible that packets from one
resource will be intermingled with packets from another resource, sent
to the same group and port. In this case the combination of the SSRC
and timestamp can be used to demultiplex the resource stream.
5. HTTP Payload format
HTTP payloads consist of a preamble header, a MIME-like header, and
entity-body content.
The purpose of the preamble header is identify the resource being
sent. Depending on the caching implementation, partially received
resources will either be discarded, or will be appropriately marked
for storage in the cache. This will allow the cache to request the
missing portion of the resource if a cache hit occurs.
Since this specification is primarily intended for use in situations
where a local repair may not be timely (such as with a frequently
updated stock ticker), where the resource will be re-multicast, or
where error correction information is transmitted on a separate group,
no repair mechanism is required.
While MIME-like headers could be used for resource identification (to
provide information such as the URI, referring URI, content length and
content range), prepending of a binary header reduces the amount of
parsing required to get at this critical information. The use of a
preamble header has the additional benefit of not requiring MIME-like
headers to be present other than in the first packet. Thus, subsequent
packets will typically contain only the preamble header and body con-
tent.
Thus, rather than adding MIME-like headers, it is intended that
senders will simply retransmit the MIME-like header obtained in the
HTTP GET response. Similarly, it is expected that the entity-body con-
tent will be retransmitted without modification.
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5.1. Preamble header
The preamble header is comprised of an 8 octet fixed portion, and a
single variable-length string.
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 2
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
| Offset (8 octets) |
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| GET request...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| GET request.... |0|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
5.1.1. Offset
The offset field is 64-bits long. For a data packet, it represents the
octet offset within the resource identified by the RTP timestamp.
Although the offset field will typically increase with increasing
sequence numbers, this need not always be the case, since portions of
a resource may be transmitted out of order. Receivers should therefore
be prepared to handle packet ordering based on the offset rather than
the sequence number.
While the offset information could have also been provided using the
Content-range: HTTP response header, this would have required inser-
tion of a MIME-like header within each packet, and parsing of this
header by the receiver. It is felt that the offset mechanism is more
efficient.
5.1.2. GET request
The GET request field is a null-terminated string, representing the
full text of the GET request that caused the resource to be transmit-
ted. At a minimum, the GET request, defined in [8], includes the GET
method, the URI and the HTTP version. While it begins on a 32-bit
boundary, the GET request field is not padded to such a boundary.
Given the expected size of the GET request field, this variable-length
string is expected to comprise the majority of the encapsulation over-
head. Given that the string may be 40 octets or larger, when the 40
octets of IP/UDP/RTP header are added, the result may be 80 octets or
more of overhead. This level of overhead would be unacceptable if it
were present in every packet.
The GET request, offset, and packet length uniquely identify the por-
tion of the resource being transmitted. Since receivers may join late,
or miss portions of the transmission, receivers must be able to
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quickly bind a timestamp to a GET request so that the incoming
resource and missing portions can be identified.
However, it is believed that this goal can be accomplished by placing
the GET request within the first packet of a series, and then only
within every subsequent nth packet. This results in a substantial
reduction in overhead. For the purposes of this specification, it is
suggested that n=4.
For packets in which the GET request is not included, a single null
octet is used to indicate the missing field.
6. Resource length
The resource length is not included in the preamble header. Typically,
this information is included within the first packet via the Content-
length: header. Although it is possible that the first and/or final
packet will be lost, we do not believe that the resource length justi-
fies inclusion within the preamble header. This is because it is not
necessary to know the total length of the resource to make a partial
GET request for the remainder of the resource should a cache hit
occur.
Note that it is possible that the final packets of a resource are
lost. In this case, the receiver will note that it has not yet
received a packet with the marker bit indicating completion of the
resource transmission, after waiting a suitable interval after recep-
tion of the last packet. After this interval has expired, the receiver
will write the incomplete resource out to the disk.
7. Non-RTP means
In addition to information transmitted within the RTP encoding, it is
expected that receivers will make use of session information transmit-
ted by non-RTP means.
For example, the existence of a data-carousel style session is
expected to be determined via SDP, transmitted in one of its encapsu-
lations, such as SAP. The SDP announcement will provide information on
the bandwidth allocated to the session, as well as the group address
(or in the case of a multi-group session, addresses) allocated to the
session.
The SDP announcement will also be expected to indicate whether the
data carousel transmission provides for re-multicast of the same
resource (in which case receivers can make partial GET requests for
the missing portion), or whether it provides continually updated
information (in which case receivers missing a portion of the resource
should make a GET request for the entire resource).
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8. Layered encoding
Reference [4] discusses use of RTP in layered multicasting, and pro-
poses guidelines for group address and port allocation, as well as
modifications to RTP semantics suitable for allocation of SSRC identi-
fiers across layered streams. SDES packets are then sent only on the
lowest layer. It is expected that these modifications, once available,
should be applicable to transmission of layered HTTP payloads.
9. Acknowledgements
Thanks to Peter Parnes of the University of Lulea, and Jim Gemmell,
Paul Leach and Thomas Pfenning of Microsoft for useful discussions of
this problem space.
10. References
[1] R. Braden. "Requirements for Internet hosts - application and
support." STD 3, RFC 1123, IETF, October 1989.
[2] H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson. "RTP: A
Transport Protocol for Real-Time Applications." RFC 1889, GMD Fokus,
January 1996.
[3] H. Schulzrinne. "RTP Profile for Audio and Video Conferences
with Minimal Control." RFC 1890, GMD Fokus, January 1996.
[4] M. F. Speer, S. McCanne. "RTP Usage with Layered Multimedia
Streams." draft-speer-avt-layered-video-01.txt, Sun Microsystems,
LBNL, June 1996.
[5] J. Palme, A. Hopmann. "MIME E-mail Encapsulation of Aggregate
Documents, such as HTML (MHTML)." draft-ietf-mhtml-spec-03.txt,
Stockholm University/KTH, ResNova Software, August, 1996.
[6] E. Levinson. "The MIME Multipart/Related Content-type." draft-
ietf-mhtml-related-00.txt, Xison, May, 1996.
[7] J. Palme. "Sending HTML in E-mail, an informational supplement."
draft-ietf-mhtml-info-03.txt, Stockholm University/KTH, August, 1996.
[8] R. Fielding, et al. "Hypertext Transfer Protocol - HTTP/1.1."
draft-ietf-http-v11-spec-07, UC Irvine, August, 1996.
[9] J. Crowcroft, Z. Wang, A. Ghosh, C. Diot. "RMFP: A Reliable Mul-
ticast Framing Protocol." draft-crowcroft-rmfp-00.txt, UCL, November,
1996.
[10] P. Parnes. "RTP Extension for Scalable Reliable Multicast."
draft-parnes-rtp-ext-srm-01.txt, LuTH/CDT, November, 1996.
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11. Authors' Addresses
Bernard Aboba
Microsoft Corporation
One Microsoft Way
Redmond, WA 98052
Phone: 206-936-6605
EMail: bernarda@microsoft.com
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