| Rfc | 4588 |
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| Title | RTP Retransmission Payload Format |
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| Author | J. Rey, D. Leon, A. Miyazaki, V.
Varsa, R. Hakenberg |
|---|
| Date | July 2006 |
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| Format: | TXT, HTML |
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| Status: | PROPOSED STANDARD |
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|
Network Working Group J. Rey
Request for Comments: 4588 Panasonic
Category: Standards Track D. Leon
Consultant
A. Miyazaki
Panasonic
V. Varsa
Nokia
R. Hakenberg
Panasonic
July 2006
RTP Retransmission Payload Format
Status of This Memo
This document specifies an Internet standards track protocol for the
Internet community, and requests discussion and suggestions for
improvements. Please refer to the current edition of the "Internet
Official Protocol Standards" (STD 1) for the standardization state
and status of this protocol. Distribution of this memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (2006).
Abstract
RTP retransmission is an effective packet loss recovery technique for
real-time applications with relaxed delay bounds. This document
describes an RTP payload format for performing retransmissions.
Retransmitted RTP packets are sent in a separate stream from the
original RTP stream. It is assumed that feedback from receivers to
senders is available. In particular, it is assumed that Real-time
Transport Control Protocol (RTCP) feedback as defined in the extended
RTP profile for RTCP-based feedback (denoted RTP/AVPF) is available
in this memo.
Table of Contents
1. Introduction ....................................................3
2. Terminology .....................................................3
3. Requirements and Design Rationale for a Retransmission Scheme ...4
3.1. Multiplexing Scheme Choice .................................6
4. Retransmission Payload Format ...................................7
5. Association of Retransmission and Original Streams ..............9
5.1. Retransmission Session Sharing .............................9
5.2. CNAME Use ..................................................9
5.3. Association at the Receiver ................................9
6. Use with the Extended RTP Profile for RTCP-based Feedback ......11
6.1. RTCP at the Sender ........................................11
6.2. RTCP Receiver Reports .....................................11
6.3. Retransmission Requests ...................................12
6.4. Timing Rules ..............................................13
7. Congestion Control .............................................13
8. Retransmission Payload Format MIME Type Registration ...........15
8.1. Introduction ..............................................15
8.2. Registration of audio/rtx .................................16
8.3. Registration of video/rtx .................................17
8.4. Registration of text/rtx ..................................18
8.5. Registration of application/rtx ...........................19
8.6. Mapping to SDP ............................................20
8.7. SDP Description with Session-Multiplexing .................20
8.8. SDP Description with SSRC-Multiplexing ....................21
9. RTSP Considerations ............................................22
9.1. RTSP Control with SSRC-Multiplexing .......................22
9.2. RTSP Control with Session-Multiplexing ....................22
9.3. RTSP Control of the Retransmission Stream .................23
9.4. Cache Control .............................................23
10. Implementation Examples .......................................23
10.1. A Minimal Receiver Implementation Example ................24
10.2. Retransmission of Layered Encoded Media in Multicast .....25
11. IANA Considerations ...........................................26
12. Security Considerations .......................................26
13. Acknowledgements ..............................................27
14. References ....................................................27
14.1. Normative References .....................................27
14.2. Informative References ...................................28
Appendix A. How to Control the Number of Rtxs. per Packet .........29
1. Introduction
Packet losses between an RTP sender and receiver may significantly
degrade the quality of the received media. Several techniques, such
as forward error correction (FEC), retransmissions, or interleaving,
may be considered to increase packet loss resiliency. RFC 2354 [8]
discusses the different options.
When choosing a repair technique for a particular application, the
tolerable latency of the application has to be taken into account.
In the case of multimedia conferencing, the end-to-end delay has to
be at most a few hundred milliseconds in order to guarantee
interactivity, which usually excludes the use of retransmission.
With sufficient latency, the efficiency of the repair scheme can be
increased. The sender may use the receiver feedback in order to
react to losses before their playout time at the receiver.
In the case of multimedia streaming, the user can tolerate an initial
latency as part of the session set-up and thus an end-to-end delay of
several seconds may be acceptable. RTP retransmission as defined in
this document is targeted at such applications.
Furthermore, the RTP retransmission method defined herein is
applicable to unicast and (small) multicast groups. The present
document defines a payload format for retransmitted RTP packets and
provides protocol rules for the sender and the receiver involved in
retransmissions.
This retransmission payload format was designed for use with the
extended RTP profile for RTCP-based feedback, AVPF [1]. It may also
be used with other RTP profiles defined in the future.
The AVPF profile allows for more frequent feedback and for early
feedback. It defines a general-purpose feedback message, i.e., NACK,
as well as codec and application-specific feedback messages. See [1]
for details.
2. Terminology
The following terms are used in this document:
CSRC: contributing source. See [3].
Original packet: an RTP packet that carries user data sent for the
first time by an RTP sender.
Original stream: the RTP stream of original packets.
Retransmission packet: an RTP packet that is to be used by the
receiver instead of a lost original packet. Such a retransmission
packet is said to be associated with the original RTP packet.
Retransmission request: a means by which an RTP receiver is able to
request that the RTP sender should send a retransmission packet for a
given original packet. Usually, an RTCP NACK packet as specified in
[1] is used as retransmission request for lost packets.
Retransmission stream: the stream of retransmission packets
associated with an original stream.
Session-multiplexing: scheme by which the original stream and the
associated retransmission stream are sent into two different RTP
sessions.
SSRC: synchronization source. See [3].
SSRC-multiplexing: scheme by which the original stream and the
retransmission stream are sent in the same RTP session with different
SSRC values.
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [2].
3. Requirements and Design Rationale for a Retransmission Scheme
The use of retransmissions in RTP as a repair method for streaming
media is appropriate in those scenarios with relaxed delay bounds and
where full reliability is not a requirement. More specifically, RTP
retransmission allows one to trade off reliability vs. delay; i.e.,
the endpoints may give up retransmitting a lost packet after a given
buffering time has elapsed. Unlike TCP, there is thus no head-of-
line blocking caused by RTP retransmissions. The implementer should
be aware that in cases where full reliability is required or higher
delay and jitter can be tolerated, TCP or other transport options
should be considered.
The RTP retransmission scheme defined in this document is designed to
fulfill the following set of requirements:
1. It must not break general RTP and RTCP mechanisms.
2. It must be suitable for unicast and small multicast groups.
3. It must work with mixers and translators.
4. It must work with all known payload types.
5. It must not prevent the use of multiple payload types in a
session.
6. In order to support the largest variety of payload formats, the
RTP receiver must be able to derive how many and which RTP packets
were lost as a result of a gap in received RTP sequence numbers.
This requirement is referred to as sequence number preservation.
Without such a requirement, it would be impossible to use
retransmission with payload formats, such as conversational text
[9] or most audio/video streaming applications, that use the RTP
sequence number to detect lost packets.
When designing a solution for RTP retransmission, several approaches
may be considered for the multiplexing of the original RTP packets
and the retransmitted RTP packets.
One approach may be to retransmit the RTP packet with its original
sequence number and send original and retransmission packets in the
same RTP stream. The retransmission packet would then be identical
to the original RTP packet, i.e., the same header (and thus same
sequence number) and the same payload. However, such an approach is
not acceptable because it would corrupt the RTCP statistics. As a
consequence, requirement 1 would not be met. Correct RTCP statistics
require that for every RTP packet within the RTP stream, the sequence
number be increased by one.
Another approach may be to multiplex original RTP packets and
retransmission packets in the same RTP stream using different payload
type values. With such an approach, the original packets and the
retransmission packets would share the same sequence number space.
As a result, the RTP receiver would not be able to infer how many and
which original packets (which sequence numbers) were lost.
In other words, this approach does not satisfy the sequence number
preservation requirement (requirement 6). This in turn implies that
requirement 4 would not be met. Interoperability with mixers and
translators would also be more difficult if they did not understand
this new retransmission payload type in a sender RTP stream. For
these reasons, a solution based on payload type multiplexing of
original packets and retransmission packets in the same RTP stream is
excluded.
Finally, the original and retransmission packets may be sent in two
separate streams. These two streams may be multiplexed either by
sending them in two different sessions , i.e., session-multiplexing,
or in the same session using different SSRC values, i.e., SSRC-
multiplexing. Since original and retransmission packets carry media
of the same type, the objections in Section 5.2 of RTP [3] to RTP
multiplexing do not apply in this case.
Mixers and translators may process the original stream and simply
discard the retransmission stream if they are unable to utilise it.
On the other hand, sending the original and retransmission packets in
two separate streams does not alone satisfy requirements 1 and 6.
For this purpose, this document includes the original sequence number
in the retransmitted packets.
In this manner, using two separate streams satisfies all the
requirements listed in this section.
3.1. Multiplexing Scheme Choice
Session-multiplexing and SSRC-multiplexing have different pros and
cons:
Session-multiplexing is based on sending the retransmission stream in
a different RTP session (as defined in RTP [3]) from that of the
original stream; i.e., the original and retransmission streams are
sent to different network addresses and/or port numbers. Having a
separate session allows more flexibility. In multicast, using two
separate sessions for the original and the retransmission streams
allows a receiver to choose whether or not to subscribe to the RTP
session carrying the retransmission stream. The original session may
also be single-source multicast while separate unicast sessions are
used to convey retransmissions to each of the receivers, which as a
result will receive only the retransmission packets they request.
The use of separate sessions also facilitates differential treatment
by the network and may simplify processing in mixers, translators,
and packet caches.
With SSRC-multiplexing, a single session is needed for the original
and the retransmission streams. This allows streaming servers and
middleware that are involved in a high number of concurrent sessions
to minimise their port usage.
This retransmission payload format allows both session-multiplexing
and SSRC-multiplexing for unicast sessions. From an implementation
point of view, there is little difference between the two approaches.
Hence, in order to maximise interoperability, both multiplexing
approaches SHOULD be supported by senders and receivers. For
multicast sessions, session-multiplexing MUST be used because the
association of the original stream and the retransmission stream is
problematic if SSRC-multiplexing is used with multicast sessions(see
Section 5.3 for motivation).
4. Retransmission Payload Format
The format of a retransmission packet is shown below:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| RTP Header |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| OSN | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| Original RTP Packet Payload |
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The RTP header usage is as follows:
In the case of session-multiplexing, the same SSRC value MUST be used
for the original stream and the retransmission stream. In the case
of an SSRC collision in either the original session or the
retransmission session, the RTP specification requires that an RTCP
BYE packet MUST be sent in the session where the collision happened.
In addition, an RTCP BYE packet MUST also be sent for the associated
stream in its own session. After a new SSRC identifier is obtained,
the SSRC of both streams MUST be set to this value.
In the case of SSRC-multiplexing, two different SSRC values MUST be
used for the original stream and the retransmission stream as
required by RTP. If an SSRC collision is detected for either the
original stream or the retransmission stream, the RTP specification
requires that an RTCP BYE packet MUST be sent for this stream. An
RTCP BYE packet MUST NOT be sent for the associated stream.
Therefore, only the stream that experienced SSRC collision MUST
choose a new SSRC value. Refer to Section 5.3 for the implications
on the original stream and retransmission stream SSRC association at
the receiver.
For either multiplexing scheme, the sequence number has the standard
definition; i.e., it MUST be one higher than the sequence number of
the preceding packet sent in the retransmission stream.
The retransmission packet timestamp MUST be set to the original
timestamp, i.e., to the timestamp of the original packet. As a
consequence, the initial RTP timestamp for the first packet of the
retransmission stream is not random but equal to the original
timestamp of the first packet that is retransmitted. See the
Security Considerations section in this document for security
implications.
Implementers have to be aware that the RTCP jitter value for the
retransmission stream does not reflect the actual network jitter
since there could be little correlation between the time a packet is
retransmitted and its original timestamp.
The payload type is dynamic. If multiple payload types using
retransmission are present in the original stream, then for each of
these, a dynamic payload type MUST be mapped to the retransmission
payload format. See Section 8.1 for the specification of how the
mapping between original and retransmission payload types is done
with Session Description Protocol (SDP).
As the retransmission packet timestamp carries the original media
timestamp, the timestamp clockrate used by the retransmission payload
type MUST be the same as the one used by the associated original
payload type. Therefore, if an RTP stream carries payload types of
different clockrates, this will also be the case for the associated
retransmission stream. Note that an RTP stream does not usually
carry payload types of different clockrates.
The payload of the RTP retransmission packet comprises the
retransmission payload header followed by the payload of the original
RTP packet. The length of the retransmission payload header is 2
octets. This payload header contains only one field, OSN (original
sequence number), which MUST be set to the sequence number of the
associated original RTP packet. The original RTP packet payload,
including any possible payload headers specific to the original
payload type, MUST be placed right after the retransmission payload
header.
For payload formats that support encoding at multiple rates, instead
of retransmitting the same payload as the original RTP packet the
sender MAY retransmit the same data encoded at a lower rate. This
aims at limiting the bandwidth usage of the retransmission stream.
When doing so, the sender MUST ensure that the receiver will still be
able to decode the payload of the already sent original packets that
might have been encoded based on the payload of the lost original
packet. In addition, if the sender chooses to retransmit at a lower
rate, the values in the payload header of the original RTP packet may
no longer apply to the retransmission packet and may need to be
modified in the retransmission packet to reflect the change in rate.
The sender SHOULD trade off the decrease in bandwidth usage with the
decrease in quality caused by resending at a lower rate.
If the original RTP header carried any profile-specific extensions,
the retransmission packet SHOULD include the same extensions
immediately following the fixed RTP header as expected by
applications running under this profile. In this case, the
retransmission payload header MUST be placed after the profile-
specific extensions.
If the original RTP header carried an RTP header extension, the
retransmission packet SHOULD carry the same header extension. This
header extension MUST be placed right after the fixed RTP header, as
specified in RTP [3]. In this case, the retransmission payload
header MUST be placed after the header extension.
If the original RTP packet contained RTP padding, that padding MUST
be removed before constructing the retransmission packet. If padding
of the retransmission packet is needed, padding MUST be performed as
with any RTP packets and the padding bit MUST be set.
The marker bit (M), the CSRC count (CC), and the CSRC list of the
original RTP header MUST be copied "as is" into the RTP header of the
retransmission packet.
5. Association of Retransmission and Original Streams
5.1. Retransmission Session Sharing
In the case of session-multiplexing, a retransmission session MUST
map to exactly one original session; i.e., the same retransmission
session cannot be used for different original sessions.
If retransmission session sharing were allowed, it would be a problem
for receivers, since they would receive retransmissions for original
sessions they might not have joined. For example, a receiver wishing
to receive only audio would receive also retransmitted video packets
if an audio and video session shared the same retransmission session.
5.2. CNAME Use
In both the session-multiplexing and the SSRC-multiplexing cases, a
sender MUST use the same RTCP CNAME [3] for an original stream and
its associated retransmission stream.
5.3. Association at the Receiver
A receiver receiving multiple original and retransmission streams
needs to associate each retransmission stream with its original
stream. The association is done differently depending on whether
session-multiplexing or SSRC-multiplexing is used.
If session-multiplexing is used, the receiver associates the two
streams having the same SSRC in the two sessions. Note that the
payload type field cannot be used to perform the association as
several media streams may have the same payload type value. The two
sessions are themselves associated out-of-band. See Section 8 for
how the grouping of the two sessions is done with SDP.
If SSRC-multiplexing is used, the receiver should first of all look
for two streams that have the same CNAME in the session. In some
cases, the CNAME may not be enough to determine the association as
multiple original streams in the same session may share the same
CNAME. For example, there can be in the same video session multiple
video streams mapping to different SSRCs and still using the same
CNAME and possibly the same payload type (PT) values. Each (or some)
of these streams may have an associated retransmission stream.
In this case, in order to find out the association between original
and retransmission streams having the same CNAME, the receiver SHOULD
behave as follows.
The association can generally be resolved when the receiver receives
a retransmission packet matching a retransmission request that had
been sent earlier. Upon reception of a retransmission packet whose
original sequence number has been previously requested, the receiver
can derive that the SSRC of the retransmission packet is associated
to the sender SSRC from which the packet was requested.
However, this mechanism might fail if there are two outstanding
requests for the same packet sequence number in two different
original streams of the session. Note that since the initial packet
sequence numbers are random, the probability of having two
outstanding requests for the same packet sequence number would be
very small. Nevertheless, in order to avoid ambiguity in the unicast
case, the receiver MUST NOT have two outstanding requests for the
same packet sequence number in two different original streams before
the association is resolved. In multicast, this ambiguity cannot be
completely avoided, because another receiver may have requested the
same sequence number from another stream. Therefore, SSRC-
multiplexing MUST NOT be used in multicast sessions.
If the receiver discovers that two senders are using the same SSRC or
if it receives an RTCP BYE packet, it MUST stop requesting
retransmissions for that SSRC. Upon reception of original RTP
packets with a new SSRC, the receiver MUST perform the SSRC
association again as described in this section.
6. Use with the Extended RTP Profile for RTCP-based Feedback
This section gives general hints for the usage of this payload format
with the extended RTP profile for RTCP-based feedback, denoted AVPF
[1]. Note that the general RTCP send and receive rules and the RTCP
packet format as specified in RTP apply, except for the changes that
the AVPF profile introduces. In short, the AVPF profile relaxes the
RTCP timing rules and specifies additional general-purpose RTCP
feedback messages. See [1] for details.
6.1. RTCP at the Sender
In the case of session-multiplexing, Sender Report (SR) packets for
the original stream are sent in the original session and SR packets
for the retransmission stream are sent in the retransmission session
according to the rules of RTP.
In the case of SSRC-multiplexing, SR packets for both original and
retransmission streams are sent in the same session according to the
rules of RTP. The original and retransmission streams are seen, as
far as the RTCP bandwidth calculation is concerned, as independent
senders belonging to the same RTP session and are thus equally
sharing the RTCP bandwidth assigned to senders.
Note that in both cases, session- and SSRC-multiplexing, BYE packets
MUST still be sent for both streams as specified in RTP. In other
words, it is not enough to send BYE packets for the original stream
only.
6.2. RTCP Receiver Reports
In the case of session-multiplexing, the receiver will send report
blocks for the original stream and the retransmission stream in
separate Receiver Report (RR) packets belonging to separate RTP
sessions. RR packets reporting on the original stream are sent in
the original RTP session while RR packets reporting on the
retransmission stream are sent in the retransmission session. The
RTCP bandwidth for these two sessions may be chosen independently
(e.g., through RTCP bandwidth modifiers [4]).
In the case of SSRC-multiplexing, the receiver sends report blocks
for the original and the retransmission streams in the same RR packet
since there is a single session.
6.3. Retransmission Requests
The NACK feedback message format defined in the AVPF profile SHOULD
be used by receivers to send retransmission requests. Whether or not
a receiver chooses to request a packet is an implementation issue.
An actual receiver implementation should take into account such
factors as the tolerable application delay, the network environment,
and the media type.
The receiver should generally assess whether the retransmitted packet
would still be useful at the time it is received. The timestamp of
the missing packet can be estimated from the timestamps of packets
preceding and/or following the sequence number gap caused by the
missing packet in the original stream. In most cases, some form of
linear estimate of the timestamp is good enough.
Furthermore, a receiver should compute an estimate of the round-trip
time (RTT) to the sender. This can be done, for example, by
measuring the retransmission delay to receive a retransmission packet
after a NACK has been sent for that packet. This estimate may also
be obtained from past observations, RTCP report round-trip time if
available, or any other means. A standard mechanism for the receiver
to estimate the RTT is specified in "RTP Control Protocol Extended
Reports (RTCP XR)" [11].
The receiver should not send a retransmission request as soon as it
detects a missing sequence number but should add some extra delay to
compensate for packet reordering. This extra delay may, for example,
be based on past observations of the experienced packet reordering.
It should be noted that, in environments where packet reordering is
rare or does not take place, e.g., if the underlying datalink layer
affords ordered delivery, the delay may be extremely low or even take
the value zero. In such cases, an appropriate "reorder delay"
algorithm may not actually be timer based, but packet based. For
example, if n number of packets are received after a gap is detected,
then it may be assumed that the packet was truly lost rather than out
of order. This may turn out to be far easier to code on some
platforms as a very short fixed FIFO packet buffer as opposed to the
timer-based mechanism.
To increase the robustness to the loss of a NACK or of a
retransmission packet, a receiver may send a new NACK for the same
packet. This is referred to as multiple retransmissions. Before
sending a new NACK for a missing packet, the receiver should rely on
a timer to be reasonably sure that the previous retransmission
attempt has failed and so avoid unnecessary retransmissions. The
timer value shall be based on the observed round-trip time. A static
or an adaptive value MAY be used. For example, an adaptive timer
could be one that changes its value with every new request for the
same packet. This document does not provide any guidelines as to how
this adaptive value should be calculated because no experiments have
been done to find this out.
NACKs MUST be sent only for the original RTP stream. Otherwise, if a
receiver wanted to perform multiple retransmissions by sending a NACK
in the retransmission stream, it would not be able to know the
original sequence number and a timestamp estimation of the packet it
requests.
Appendix A gives some guidelines as to how to control the number of
retransmissions.
6.4. Timing Rules
The NACK feedback message may be sent in a regular full compound RTCP
packet or in an early RTCP packet, as per AVPF [1]. Sending a NACK
in an early packet allows reacting more quickly to a given packet
loss. However, in that case if a new packet loss occurs right after
the early RTCP packet was sent, the receiver will then have to wait
for the next regular RTCP compound packet after the early packet.
Sending NACKs only in regular RTCP compound decreases the maximum
delay between detecting an original packet loss and being able to
send a NACK for that packet. Implementers should consider the
possible implications of this fact for the application being used.
Furthermore, receivers may make use of the minimum interval between
regular RTCP compound packets. This interval can be used to keep
regular receiver reporting down to a minimum, while still allowing
receivers to send early RTCP packets during periods requiring more
frequent feedback, e.g., times of higher packet loss rate. Note that
although RTCP packets may be suppressed because they do not contain
NACKs, the same RTCP bandwidth as if they were sent needs to be
available. See AVPF [1] for details on the use of the minimum
interval.
7. Congestion Control
RTP retransmission poses a risk of increasing network congestion. In
a best-effort environment, packet loss is caused by congestion.
Reacting to loss by retransmission of older data without decreasing
the rate of the original stream would thus further increase
congestion. Implementations SHOULD follow the recommendations below
in order to use retransmission.
The RTP profile under which the retransmission scheme is used defines
an appropriate congestion control mechanism in different
environments. Following the rules under the profile, an RTP
application can determine its acceptable bitrate and packet rate in
order to be fair to other TCP or RTP flows.
If an RTP application uses retransmission, the acceptable packet rate
and bitrate include both the original and retransmitted data. This
guarantees that an application using retransmission achieves the same
fairness as one that does not. Such a rule would translate in
practice into the following actions:
If enhanced service is used, it should be made sure that the total
bitrate and packet rate do not exceed that of the requested service.
It should be further monitored that the requested services are
actually delivered. In a best-effort environment, the sender SHOULD
NOT send retransmission packets without reducing the packet rate and
bitrate of the original stream (for example, by encoding the data at
a lower rate).
In addition, the sender MAY selectively retransmit only the packets
that it deems important and ignore NACK messages for other packets in
order to limit the bitrate.
These congestion control mechanisms should keep the packet loss rate
within acceptable parameters. In the context of congestion control,
packet loss is considered acceptable if a TCP flow across the same
network path and experiencing the same network conditions would
achieve, on a reasonable timescale, an average throughput that is not
less than the one the RTP flow achieves. If congestion is not kept
under control, then retransmission SHOULD NOT be used.
Retransmissions MAY still be sent in some cases, e.g., in wireless
links where packet losses are not caused by congestion, if the server
(or the client that makes the retransmission request) estimates that
a particular packet or frame is important to continue play out, or if
an RTSP PAUSE has been issued to allow the buffer to fill up (RTSP
PAUSE does not affect the sending of retransmissions).
Finally, it may further be necessary to adapt the transmission rate
(or the number of layers subscribed for a layered multicast session),
or to arrange for the receiver to leave the session.
8. Retransmission Payload Format MIME Type Registration
8.1. Introduction
The following MIME subtype name and parameters are introduced in this
document: "rtx", "rtx-time", and "apt".
The binding used for the retransmission stream to the payload type
number is indicated by an rtpmap attribute. The MIME subtype name
used in the binding is "rtx".
The "apt" (associated payload type) parameter MUST be used to map the
retransmission payload type to the associated original stream payload
type. If multiple original payload types are used, then multiple
"apt" parameters MUST be included to map each original payload type
to a different retransmission payload type.
An OPTIONAL payload-format-specific parameter, "rtx-time", indicates
the maximum time a sender will keep an original RTP packet in its
buffers available for retransmission. This time starts with the
first transmission of the packet.
The syntax is as follows:
a=fmtp:<number> apt=<apt-value>;rtx-time=<rtx-time-val>
where
<number>: indicates the dynamic payload type number assigned to
the retransmission payload format in an rtpmap attribute.
<apt-value>: is the value of the original stream payload type to
which this retransmission stream payload type is associated.
<rtx-time-val>: specifies the time in milliseconds (measured from
the time a packet was first sent) that a sender keeps an RTP
packet in its buffers available for retransmission. The absence
of the rtx-time parameter for a retransmission stream means that
the maximum retransmission time is not defined, but MAY be
negotiated by other means.
8.2. Registration of audio/rtx
MIME type: audio
MIME subtype: rtx
Required parameters:
rate: the RTP timestamp clockrate is equal to the RTP timestamp
clockrate of the media that is retransmitted.
apt: associated payload type. The value of this parameter is the
payload type of the associated original stream.
Optional parameters:
rtx-time: indicates the time in milliseconds (measured from the
time a packet was first sent) that the sender keeps an RTP packet
in its buffers available for retransmission.
Encoding considerations: this type is only defined for transfer via
RTP.
Security considerations: see Section 12 of RFC 4588
Interoperability considerations: none
Published specification: RFC 4588
Applications which use this media type: multimedia streaming
applications
Additional information: none
Person & email address to contact for further information:
jose.rey@eu.panasonic.com
davidleon123@yahoo.com
avt@ietf.org
Intended usage: COMMON
Authors:
Jose Rey
David Leon
Change controller:
IETF AVT WG delegated from the IESG
8.3. Registration of video/rtx
MIME type: video
MIME subtype: rtx
Required parameters:
rate: the RTP timestamp clockrate is equal to the RTP timestamp
clockrate of the media that is retransmitted.
apt: associated payload type. The value of this parameter is the
payload type of the associated original stream.
Optional parameters:
rtx-time: indicates the time in milliseconds (measured from the
time a packet was first sent) that the sender keeps an RTP packet
in its buffers available for retransmission.
Encoding considerations: this type is only defined for transfer via
RTP.
Security considerations: see Section 12 of RFC 4588
Interoperability considerations: none
Published specification: RFC 4588
Applications which use this media type: multimedia streaming
applications
Additional information: none
Person & email address to contact for further information:
jose.rey@eu.panasonic.com
davidleon123@yahoo.com
avt@ietf.org
Intended usage: COMMON
Authors:
Jose Rey
David Leon
Change controller:
IETF AVT WG delegated from the IESG
8.4. Registration of text/rtx
MIME type: text
MIME subtype: rtx
Required parameters:
rate: the RTP timestamp clockrate is equal to the RTP timestamp
clockrate of the media that is retransmitted.
apt: associated payload type. The value of this parameter is the
payload type of the associated original stream.
Optional parameters:
rtx-time: indicates the time in milliseconds (measured from the
time a packet was first sent) that the sender keeps an RTP packet
in its buffers available for retransmission.
Encoding considerations: this type is only defined for transfer via
RTP.
Security considerations: see Section 12 of RFC 4588
Interoperability considerations: none
Published specification: RFC 4588
Applications which use this media type: multimedia streaming
applications
Additional information: none
Person & email address to contact for further information:
jose.rey@eu.panasonic.com
davidleon123@yahoo.com
avt@ietf.org
Intended usage: COMMON
Authors:
Jose Rey
David Leon
Change controller:
IETF AVT WG delegated from the IESG
8.5. Registration of application/rtx
MIME type: application
MIME subtype: rtx
Required parameters:
rate: the RTP timestamp clockrate is equal to the RTP timestamp
clockrate of the media that is retransmitted.
apt: associated payload type. The value of this parameter is the
payload type of the associated original stream.
Optional parameters:
rtx-time: indicates the time in milliseconds (measured from the
time a packet was first sent) that the sender keeps an RTP packet
in its buffers available for retransmission.
Encoding considerations: this type is only defined for transfer via
RTP.
Security considerations: see Section 12 of RFC 4588
Interoperability considerations: none
Published specification: RFC 4588
Applications which use this media type: multimedia streaming
applications
Additional information: none
Person & email address to contact for further information:
jose.rey@eu.panasonic.com
davidleon123@yahoo.com
avt@ietf.org
Intended usage: COMMON
Authors:
Jose Rey
David Leon
Change controller:
IETF AVT WG delegated from the IESG
8.6. Mapping to SDP
The information carried in the MIME media type specification has a
specific mapping to fields in SDP [5], which is commonly used to
describe RTP sessions. When SDP is used to specify retransmissions
for an RTP stream, the mapping is done as follows:
- The MIME types ("video"), ("audio"), ("text"), and ("application")
go in the SDP "m=" as the media name.
- The MIME subtype ("rtx") goes in SDP "a=rtpmap" as the encoding
name. The RTP clockrate in "a=rtpmap" MUST be that of the
retransmission payload type. See Section 4 for details on this.
- The AVPF profile-specific parameters "ack" and "nack" go in SDP
"a=rtcp-fb". Several SDP "a=rtcp-fb" are used for several types
of feedback. See the AVPF profile [1] for details.
- The retransmission payload-format-specific parameters "apt" and
"rtx-time" go in the SDP "a=fmtp" as a semicolon-separated list of
parameter=value pairs.
- Any remaining parameters go in the SDP "a=fmtp" attribute by
copying them directly from the MIME media type string as a
semicolon-separated list of parameter=value pairs.
In the following sections, some example SDP descriptions are
presented. In some of these examples, long lines are folded to meet
the column width constraints of this document; the backslash ("\") at
the end of a line and the carriage return that follows it should be
ignored.
8.7. SDP Description with Session-Multiplexing
In the case of session-multiplexing, the SDP description contains one
media specification "m" line per RTP session. The SDP MUST provide
the grouping of the original and associated retransmission sessions'
"m" lines, using the Flow Identification (FID) semantics defined in
RFC 3388 [6].
The following example specifies two original, AMR and MPEG-4, streams
on ports 49170 and 49174 and their corresponding retransmission
streams on ports 49172 and 49176, respectively:
v=0
o=mascha 2980675221 2980675778 IN IP4 host.example.net
c=IN IP4 192.0.2.0
a=group:FID 1 2
a=group:FID 3 4
m=audio 49170 RTP/AVPF 96
a=rtpmap:96 AMR/8000
a=fmtp:96 octet-align=1
a=rtcp-fb:96 nack
a=mid:1
m=audio 49172 RTP/AVPF 97
a=rtpmap:97 rtx/8000
a=fmtp:97 apt=96;rtx-time=3000
a=mid:2
m=video 49174 RTP/AVPF 98
a=rtpmap:98 MP4V-ES/90000
a=rtcp-fb:98 nack
a=fmtp:98 profile-level-id=8;config=01010000012000884006682C209\
0A21F
a=mid:3
m=video 49176 RTP/AVPF 99
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=98;rtx-time=3000
a=mid:4
A special case of the SDP description is a description that contains
only one original session "m" line and one retransmission session "m"
line, the grouping is then obvious and FID semantics MAY be omitted
in this special case only.
This is illustrated in the following example, which is an SDP
description for a single original MPEG-4 stream and its corresponding
retransmission session:
v=0
o=mascha 2980675221 2980675778 IN IP4 host.example.net
c=IN IP4 192.0.2.0
m=video 49170 RTP/AVPF 96
a=rtpmap:96 MP4V-ES/90000
a=rtcp-fb:96 nack
a=fmtp:96 profile-level-id=8;config=01010000012000884006682C209\
0A21F
m=video 49172 RTP/AVPF 97
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96;rtx-time=3000
8.8. SDP Description with SSRC-Multiplexing
The following is an example of an SDP description for an RTP video
session using SSRC-multiplexing with similar parameters as in the
single-session example above:
v=0
o=mascha 2980675221 2980675778 IN IP4 host.example.net
c=IN IP4 192.0.2.0
m=video 49170 RTP/AVPF 96 97
a=rtpmap:96 MP4V-ES/90000
a=rtcp-fb:96 nack
a=fmtp:96 profile-level-id=8;config=01010000012000884006682C209\
0A21F
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96;rtx-time=3000
9. RTSP Considerations
The Real Time Streaming Protocol (RTSP), RFC 2326 [7], is an
application-level protocol for control over the delivery of data with
real-time properties. This section looks at the issues involved in
controlling RTP sessions that use retransmissions.
9.1. RTSP Control with SSRC-Multiplexing
In the case of SSRC-multiplexing, the "m" line includes both original
and retransmission payload types and has a single RTSP "control"
attribute. The receiver uses the "m" line to request SETUP and
TEARDOWN of the whole media session. The RTP profile contained in
the Transport header MUST be the AVPF profile or another suitable
profile allowing extended feedback. If the SSRC value is included in
the SETUP response's Transport header, it MUST be that of the
original stream.
In order to control the sending of the session original media stream,
the receiver sends as usual PLAY and PAUSE requests to the sender for
the session. The RTP-info header that is used to set RTP-specific
parameters in the PLAY response MUST be set according to the RTP
information of the original stream.
When the receiver starts receiving the original stream, it can then
request retransmission through RTCP NACKs without additional RTSP
signalling.
9.2. RTSP Control with Session-Multiplexing
In the case of session-multiplexing, each SDP "m" line has an RTSP
"control" attribute. Hence, when retransmission is used, both the
original session and the retransmission have their own "control"
attributes. The receiver can associate the original session and the
retransmission session through the FID semantics as specified in
Section 8.
The original and the retransmission streams are set up and torn down
separately through their respective media "control" attribute. The
RTP profile contained in the Transport header MUST be the AVPF
profile or another suitable profile allowing extended feedback for
both the original and the retransmission sessions.
The RTSP presentation SHOULD support aggregate control and SHOULD
contain a session-level RTSP URL. The receiver SHOULD use aggregate
control for an original session and its associated retransmission
session. Otherwise, there would need to be two different 'session-
id' values, i.e., different values for the original and
retransmission sessions, and the sender would not know how to
associate them.
The session-level "control" attribute is then used as usual to
control the playing of the original stream. When the receiver starts
receiving the original stream, it can then request retransmissions
through RTCP without additional RTSP signalling.
9.3. RTSP Control of the Retransmission Stream
Because of the nature of retransmissions, the sending of
retransmission packets SHOULD NOT be controlled through RTSP PLAY and
PAUSE requests. The PLAY and PAUSE requests SHOULD NOT affect the
retransmission stream. Retransmission packets are sent upon receiver
requests in the original RTCP stream, regardless of the state.
9.4. Cache Control
Retransmission streams SHOULD NOT be cached.
In the case of session-multiplexing, the "Cache-Control" header
SHOULD be set to "no-cache" for the retransmission stream.
In the case of SSRC-multiplexing, RTSP cannot specify independent
caching for the retransmission stream, because there is a single "m"
line in SDP. Therefore, the implementer should take this fact into
account when deciding whether or not to cache an SSRC-multiplexed
session.
10. Implementation Examples
This document mandates only the sender and receiver behaviours that
are necessary for interoperability. In addition, certain algorithms,
such as rate control or buffer management when targeted at specific
environments, may enhance the retransmission efficiency.
This section gives an overview of different implementation options
allowed within this specification.
The first example describes a minimal receiver implementation. With
this implementation, it is possible to retransmit lost RTP packets,
detect efficiently the loss of retransmissions, and perform multiple
retransmissions, if needed. Most of the necessary processing is done
at the server.
The second example shows how retransmissions may be used in (small)
multicast groups in conjunction with layered encoding. It
illustrates that retransmissions and layered encoding may be
complementary techniques.
10.1. A Minimal Receiver Implementation Example
This section gives an example of an implementation supporting
multiple retransmissions. The sender transmits the original data in
RTP packets using the MPEG-4 video RTP payload format. It is assumed
that NACK feedback messages are used, as per [1]. An SDP description
example with SSRC-multiplexing is given below:
v=0
o=mascha 2980675221 2980675778 IN IP4 host.example.net
c=IN IP4 192.0.2.0
m=video 49170 RTP/AVPF 96 97
a=rtpmap:96 MP4V-ES/90000
a=rtcp-fb:96 nack
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96;rtx-time=3000
The format-specific parameter "rtx-time" indicates that the server
will buffer the sent packets in a retransmission buffer for 3.0
seconds, after which the packets are deleted from the retransmission
buffer and will never be sent again.
In this implementation example, the required RTP receiver processing
to handle retransmission is kept to a minimum. The receiver detects
packet loss from the gaps observed in the received sequence numbers.
It signals lost packets to the sender through NACKs as defined in the
AVPF profile [1]. The receiver should take into account the
signalled sender retransmission buffer length in order to dimension
its own reception buffer. It should also derive from the buffer
length the maximum number of times the retransmission of a packet can
be requested.
The sender should retransmit the packets selectively; i.e., it should
choose whether to retransmit a requested packet depending on the
packet importance, the observed Quality of Service (QoS), and
congestion state of the network connection to the receiver.
Obviously, the sender processing increases with the number of
receivers as state information and processing load must be allocated
to each receiver.
10.2. Retransmission of Layered Encoded Media in Multicast
This section shows how to combine retransmissions with layered
encoding in multicast sessions. Note that the retransmission
framework is offered only for small multicast applications. Refer to
RFC 2887 [10] for a discussion of the problems of NACK implosion,
severe congestion caused by feedback traffic, in large-group reliable
multicast applications.
Packets of different importance are sent in different RTP sessions.
The retransmission streams corresponding to the different layers can
themselves be seen as different retransmission layers. The relative
importance of the different retransmission streams should reflect the
relative importance of the different original streams.
In multicast, SSRC-multiplexing of the original and retransmission
streams is not allowed as per Section 5.3 of this document. For this
reason, the retransmission stream(s) MUST be sent in different RTP
session(s) using session-multiplexing.
An SDP description example of multicast retransmissions for layered
encoded media is given below:
m=video 8000 RTP/AVPF 98
c=IN IP4 224.2.1.0/127/3
a=rtpmap:98 MP4V-ES/90000
a=rtcp-fb:98 nack
m=video 8000 RTP/AVPF 99
c=IN IP4 224.2.1.3/127/3
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=98;rtx-time=3000
The server and the receiver may implement the retransmission methods
illustrated in the previous examples. In addition, they may choose
to request and retransmit a lost packet depending on the layer it
belongs to.
11. IANA Considerations
A new MIME subtype name, "rtx", has been registered for four
different media types, as follows: "video", "audio", "text" and
"application". An additional REQUIRED parameter, "apt", and an
OPTIONAL parameter, "rtx-time", are defined. See Section 8 for
details.
12. Security Considerations
RTP packets using the payload format defined in this specification
are subject to the general security considerations discussed in RTP
[3], Section 9.
In common streaming scenarios message authentication, data integrity,
replay protection, and confidentiality are desired.
The absence of authentication may enable man-in-the-middle and replay
attacks, which can be very harmful for RTP retransmission. For
example: tampered RTCP packets may trigger inappropriate
retransmissions that effectively reduce the actual bitrate share
allocated to the original data stream, tampered RTP retransmission
packets could cause the client's decoder to crash, and tampered
retransmission requests may invalidate the SSRC association mechanism
described in Section 5 of this document. On the other hand, replayed
packets could lead to false reordering and RTT measurements (required
for the retransmission request strategy) and may cause the receiver
buffer to overflow.
Furthermore, in order to ensure confidentiality of the data, the
original payload data needs to be encrypted. There is actually no
need to encrypt the 2-byte retransmission payload header since it
does not provide any hints about the data content.
Furthermore, it is RECOMMENDED that the cryptography mechanisms used
for this payload format provide protection against known plaintext
attacks. RTP recommends that the initial RTP timestamp SHOULD be
random to secure the stream against known plaintext attacks. This
payload format does not follow this recommendation as the initial
timestamp will be the media timestamp of the first retransmitted
packet. However, since the initial timestamp of the original stream
is itself random, if the original stream is encrypted, the first
retransmitted packet timestamp would also be random to an attacker.
Therefore, confidentiality would not be compromised.
If cryptography is used to provide security services on the original
stream, then the same services, with equivalent cryptographic
strength, MUST be provided on the retransmission stream. The use of
the same key for the retransmitted stream and the original stream may
lead to security problems, e.g., two-time pads. Refer to Section 9.1
of the Secure Real-Time Transport Protocol (SRTP) [12] for a
discussion the implications of two-time pads and how to avoid them.
At the time of writing this document, SRTP does not provide all the
security services mentioned. There are, at least, two reasons for
this: 1) the occurrence of two-time pads and 2) the fact that this
payload format typically works under the RTP/AVPF profile whereas
SRTP only supports RTP/AVP. An adapted variant of SRTP shall solve
these shortcomings in the future.
Congestion control considerations with the use of retransmission are
dealt with in Section 7 of this document.
13. Acknowledgements
We would like to express our gratitude to Carsten Burmeister for his
participation in the development of this document. Our thanks also
go to Koichi Hata, Colin Perkins, Stephen Casner, Magnus Westerlund,
Go Hori, and Rahul Agarwal for their helpful comments.
14. References
14.1. Normative References
[1] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP profile for Real-time Transport Control Protocol
(RTCP)-Based feedback", RFC 4585, July 2006.
[2] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
[3] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
"RTP: A Transport Protocol for Real-Time Applications", STD 64,
RFC 3550, July 2003.
[4] Casner, S., "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC 3556,
July 2003.
[5] Handley, M. and V. Jacobson, "SDP: Session Description
Protocol", RFC 2327, April 1998.
[6] Camarillo, G., Eriksson, G., Holler, J., and H. Schulzrinne,
"Grouping of Media Lines in the Session Description Protocol
(SDP)", RFC 3388, December 2002.
[7] Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time Streaming
Protocol (RTSP)", RFC 2326, April 1998.
14.2. Informative References
[8] Perkins, C. and O. Hodson, "Options for Repair of Streaming
Media", RFC 2354, June 1998.
[9] Hellstrom, G. and P. Jones, "RTP Payload for Text Conversation",
RFC 4103, June 2005.
[10] Handley, M., Floyd, S., Whetten, B., Kermode, R., Vicisano, L.,
and M. Luby, "The Reliable Multicast Design Space for Bulk Data
Transfer", RFC 2887, August 2000.
[11] Friedman, T., Caceres, R., and A. Clark, "RTP Control Protocol
Extended Reports (RTCP XR)", RFC 3611, November 2003.
[12] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC
3711, March 2004.
Appendix A. How to Control the Number of Rtxs. per Packet
Finding out the number of retransmissions (rtxs.) per packet for
achieving the best possible transmission is a difficult task. Of
course, the absolute minimum should be one (1); otherwise, do not use
this payload format. Moreover, as of date of publication, the
authors were not aware of any studies on the number of
retransmissions per packet that should be used for best performance.
To help implementers and researchers on this item, this section
describes an estimate of the buffering time required for achieving a
given number of retransmissions. Once this time has been calculated,
it can be communicated to the client via SDP parameter "rtx-time", as
defined in this document.
A.1. Scenario and Assumptions
* Streaming scenario with relaxed delay bounds. Client and server
are provided with buffering space as indicated by the parameter
"rtx-time" in SDP.
* RTP AVPF profile used with SSRC-multiplexing retransmission scheme:
1 SSRC for original packets, 1 for retransmission packets.
* Default RTCP bandwidth share for SRs and RRs, i.e., SR+RR = 0.05.
We have senders (2) and receivers (1). Receivers and senders get
equally 1/3 of the RTCP bandwidth share because the proportion of
senders is greater than 1/4 of session members.
* avg-rtcp-size is approximated by 120 bytes. This is a rounded-up
average of 2 SRs, one for each SSRC, containing 40/8/28/32 bytes
for IPv6/UDP/SR/SDES with CNAME, thus making 105 bytes each; and a
RR with 40/8/64/32 bytes for IPv6/UDP/2*RR/SDES, making 157 bytes.
Since senders and receivers share the RTCP bandwidth equally, then
avg-rtcp-size = (157+105+105)/3 = 117.3 ~= 120 bytes. The
important characteristic of this value is that it is something over
100 bytes, which seems to be a representative figure for typical
configurations.
* The profile used is AVPF [1] and Generic NACKs are used for
requesting retransmissions. This adds 16 bytes of overhead for 1
NACK and 4 bytes more for every additional NACK Feedback Control
Information (FCI) field.
* We assume a worst-case scenario in which each packet exhausts its
corresponding number of available retransmissions, N, before it is
received. This means that if a packet is requested for
retransmission a maximum of 2 times, the corresponding generic NACK
report block requesting that particular packet is sent in two
consecutive RTCP compounds; likewise, if it is requested for
retransmission 10 times, then the generic NACK is sent 10 times.
This assumption makes the RTCP packet size approximately constant
after N*RTCP intervals (seconds), namely, to avg-rtcp-size = 120 +
(receiver-RTCP-bw-share)*(12 + 4*N). In our case, the receiver
RTCP bandwidth share is 1/3; thus, avg-rtcp-size = 124 + 4*N/3.
* Two delay parameters are difficult to approximate and may be
implementation dependent. Therefore, we list them here explicitly
without assigning them a particular value: one is the packet loss
detection time (T2), and the other is feedback processing and
queuing time for retransmissions (T5). Implementers shall assign
appropriate values to these two parameters.
Graphically, we have the following:
Sender
+-+---------------------------------^-----\-----------------
\ \ / \
\ \ | |
SN=0 \ \ SN=1 / \ RTX(SN=0)
\ \ / \
X \ / \
`. / \
\ / \
\ | |
\ / \ ......
\ / \
-------------V----D--------/-----------------------V--------
T1 T2 T3 T4 T5 T1 ........
Receiver
Legend:
=======
DL: downlink (client->server)
UL: uplink (server->client)
Time unit is seconds, s.
Bitrate unit is bits per second, bps.
DL transmission time: T1 = physical-delay-DL +
tx-delay-DL(=avg-pkt-size/DL-bitrate) + interarrival-delay-jitter
Time to detect packet loss: T2 = pkt-loss-detect-time
Time to report packet loss: T3 = time-to-next-rtcp-report
UL transmission time: T4 = physical-delay-UL +
transmission-delay-UL + interarrival-delay-jitter
Retransmissions processing time: T5 = feedback-processing-time +
rtx-queuing-time
A.2. Goal
To find an estimate of the buffering time, T(), that a streaming
server shall use in order to enable a given number of retransmissions
for each packet, N. This time is approximately equal at the server
and at the client, if one considers that the client starts buffering
T1 seconds later.
A.3. Solution
First, we find the value of the estimate for 1 retransmission,
T(1)=T:
T = T1 + T2 + T3 + T4 + T5
Since T1 + T4 ~= RTT,
T = RTT + T2 + T3 + T5
The worst case for T3 would be that we assume that reporting has to
wait a whole RTCP interval and that the maximum randomization factor
of 1.5 is applied. Therefore, after applying the subsequent
compensation to avoid traffic bursts (see Appendix A.7 of RTP [3]),
we have that T3 = 1.5/1.21828*RTCP-Interval. Thus,
T = RTT + 1.2312*RTCP-Interval + T2 + T5
On the other hand, RTCP-Interval = avg-rtcp-size*8*(senders +
receivers)/(RR+RS). In this scenario: sender + receivers = 3; RR+RS
is the receiver report plus sender report bandwidth share, in this
case, equal to the default 5% of session bandwidth, bw. We assume an
average RTCP packet size, avg-rtcp-size = 120 bytes. Thus:
T = RTT + 1.2312*avg-rtcp-size*8*3/(0.05*bw) + T2 + T5
for 1 retransmission.
For enabling N retransmissions, the available buffering time in a
streaming server or client is approximately:
T(N) = N*(RTT+1.2312*avg-rtcp-size*8*3/(0.05*bw) + T2 + T5)
where, as per above,
avg-rtcp-size = 120 + (receiver-RTCP-bw-share)*(12 + 4*N)
= 120 + (1/3)*(12 + 4*N)
= 124 + 4*N/3.
A.4. Numbers
If we ignore the effect of T2 and T5, i.e., assume that all losses
are detected immediately and that there is no additional delay due to
feedback processing or retransmission queuing, we have the following
buffering times for different values of N:
RTCP w/ several Generic NACKs; variable packet size = 124 + 4*N/3
bytes
|============|=====|======================================|
| RTP BW | RTT | N value |
|============|=====| 1 2 5 7 10 |
|======================================|
64000 0,05 1,21 2,44 6,28 8,97 13,18
128000 0,05 0,63 1,27 3,27 4,66 6,84
256000 0,05 0,34 0,68 1,76 2,50 3,67
512000 0,05 0,19 0,39 1,00 1,43 2,09
1024000 0,05 0,12 0,25 0,63 0,89 1,29
5000000 0,05 0,06 0,13 0,33 0,46 0,66
10000000 0,05 0,06 0,11 0,29 0,41 0,58
64000 0,2 1,36 2,74 7,03 10,02 14,68
128000 0,2 0,78 1,57 4,02 5,71 8,34
256000 0,2 0,49 0,98 2,51 3,55 5,17
512000 0,2 0,34 0,69 1,75 2,48 3,59
1024000 0,2 0,27 0,55 1,38 1,94 2,79
5000000 0,2 0,21 0,43 1,08 1,51 2,16
10000000 0,2 0,21 0,41 1,04 1,46 2,08
64000 1 2,16 4,34 11,03 15,62 22,68
128000 1 1,58 3,17 8,02 11,31 16,34
256000 1 1,29 2,58 6,51 9,15 13,17
512000 1 1,14 2,29 5,75 8,08 11,59
1024000 1 1,07 2,15 5,38 7,54 10,79
5000000 1 1,01 2,03 5,08 7,11 10,16
10000000 1 1,01 2,01 5,04 7,06 10,08
To quantify the error of not taking the Generic NACKS into account,
we can do the same numbers, but ignoring the Generic NACK
contribution, avg-rtcp-size ~= 120 bytes. As we see from below, this
may result in a buffer estimation error of 1-1.5 seconds (5-10%) for
lower bandwidth values and higher number of retransmissions. This
effect is low in this case. Nevertheless, it should be carefully
evaluated for the particular scenario; that is why the formula
includes it.
RTCP w/o Generic NACK, fixed packet size ~= 120 bytes
|============|=====|======================================|
| RTP BW | RTT | N value |
|============|=====| 1 2 5 7 10 |
|======================================|
64000 0,05 1,16 2,32 5,79 8,11 11,58
128000 0,05 0,60 1,21 3,02 4,23 6,04
256000 0,05 0,33 0,65 1,64 2,29 3,27
512000 0,05 0,19 0,38 0,94 1,32 1,89
1024000 0,05 0,12 0,24 0,60 0,83 1,19
5000000 0,05 0,06 0,13 0,32 0,45 0,64
10000000 0,05 0,06 0,11 0,29 0,40 0,57
64000 0,2 1,31 2,62 6,54 9,16 13,08
128000 0,2 0,75 1,51 3,77 5,28 7,54
256000 0,2 0,48 0,95 2,39 3,34 4,77
512000 0,2 0,34 0,68 1,69 2,37 3,39
1024000 0,2 0,27 0,54 1,35 1,88 2,69
5000000 0,2 0,21 0,43 1,07 1,50 2,14
10000000 0,2 0,21 0,41 1,04 1,45 2,07
64000 1 2,11 4,22 10,54 14,76 21,08
128000 1 1,55 3,11 7,77 10,88 15,54
256000 1 1,28 2,55 6,39 8,94 12,77
512000 1 1,14 2,28 5,69 7,97 11,39
1024000 1 1,07 2,14 5,35 7,48 10,69
5000000 1 1,01 2,03 5,07 7,10 10,14
10000000 1 1,01 2,01 5,04 7,05 10,07
Authors' Addresses
Jose Rey
Panasonic R&D Center Germany GmbH
Monzastr. 4c
D-63225 Langen, Germany
Phone: +49-6103-766-134
Fax: +49-6103-766-166
EMail: jose.rey@eu.panasonic.com
David Leon
Consultant
EMail: davidleon123@yahoo.com
Akihiro Miyazaki
Matsushita Electric Industrial Co., Ltd
1006, Kadoma, Kadoma City, Osaka, Japan
Phone: +81-6-6900-9172
Fax: +81-6-6900-9173
EMail: miyazaki.akihiro@jp.panasonic.com
Viktor Varsa
Nokia Research Center
6000 Connection Drive
Irving, TX. USA
Phone: 1-972-374-1861
EMail: viktor.varsa@nokia.com
Rolf Hakenberg
Panasonic R&D Center Germany GmbH
Monzastr. 4c
D-63225 Langen, Germany
Phone: +49-6103-766-162
Fax: +49-6103-766-166
EMail: rolf.hakenberg@eu.panasonic.com
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