AVTEXT A. Begen
Internet-Draft Cisco
Intended status: Standards Track C. Perkins
Expires: April 05, 2014 University of Glasgow
October 02, 2013
Duplicating RTP Streams
draft-ietf-avtext-rtp-duplication-04
Abstract
Packet loss is undesirable for real-time multimedia sessions, but can
occur due to congestion, or other unplanned network outages. This is
especially true for IP multicast networks, where packet loss patterns
can vary greatly between receivers. One technique that can be used
to recover from packet loss without incurring unbounded delay for all
the receivers is to duplicate the packets and send them in separate
redundant streams. This document explains how Real-time Transport
Protocol (RTP) streams can be duplicated without breaking RTP or RTP
Control Protocol (RTCP) rules.
Status of This Memo
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Terminology and Requirements Notation . . . . . . . . . . . . 3
3. Dual Streaming Use Cases . . . . . . . . . . . . . . . . . . 3
3.1. Temporal Redundancy . . . . . . . . . . . . . . . . . . . 3
3.2. Spatial Redundancy . . . . . . . . . . . . . . . . . . . 4
3.3. Dual Streaming over a Single Path or Multiple Paths . . . 4
3.4. Requirements . . . . . . . . . . . . . . . . . . . . . . 5
4. Use of RTP and RTCP with Temporal Redundancy . . . . . . . . 6
4.1. RTCP Considerations . . . . . . . . . . . . . . . . . . . 6
4.2. Signaling Considerations . . . . . . . . . . . . . . . . 6
5. Use of RTP and RTCP with Spatial Redundancy . . . . . . . . . 8
5.1. RTCP Considerations . . . . . . . . . . . . . . . . . . . 8
5.2. Signaling Considerations . . . . . . . . . . . . . . . . 8
6. Use of RTP and RTCP with Temporal and Spatial Redundancy . . 9
7. Congestion Control Considerations . . . . . . . . . . . . . . 9
8. Security Considerations . . . . . . . . . . . . . . . . . . . 10
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 11
10. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 11
11. References . . . . . . . . . . . . . . . . . . . . . . . . . 11
11.1. Normative References . . . . . . . . . . . . . . . . . . 11
11.2. Informative References . . . . . . . . . . . . . . . . . 11
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 12
1. Introduction
The Real-time Transport Protocol (RTP) [RFC3550] is widely used today
for delivering IPTV traffic, and other real-time multimedia sessions.
Many of these applications support very large numbers of receivers,
and rely on intra-domain UDP/IP multicast for efficient distribution
of traffic within the network.
While this combination has proved successful, there does exist a
weakness. As [RFC2354] noted, packet loss is not avoidable, even in
a carefully managed network. This loss might be due to congestion,
it might also be a result of an unplanned outage caused by a flapping
link, link or interface failure, a software bug, or a maintenance
person accidentally cutting the wrong fiber. Since UDP/IP flows do
not provide any means for detecting loss and retransmitting packets,
it leaves up to the RTP layer and the applications to detect, and
recover from, packet loss.
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One technique to recover from packet loss without incurring unbounded
delay for all the receivers is to duplicate the packets and send them
in separate redundant streams. Variations on this idea have been
implemented and deployed today [IC2011]. However, duplication of RTP
streams without breaking the RTP and RTCP functionality has not been
documented properly. This document discusses the most common use
cases and explains how duplication can be achieved for RTP streams in
such use cases to address the immediate market needs. In the future,
if there will be a different use case, which is not covered by this
document, a new specification that explains how RTP duplication
should be done in such a scenario may be needed.
Stream duplication offers a simple way to protect media flows from
packet loss. It has a comparatively high bandwidth overhead, since
everything is sent twice, but with a low processing overhead. It is
also very predictable in its overheads. Alternative approaches, for
example, retransmission-based recovery [RFC4588] or Forward Error
Correction [RFC6363], may be suitable in some other cases.
2. Terminology and Requirements Notation
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in
[RFC2119].
3. Dual Streaming Use Cases
Dual streaming refers to a technique that involves transmitting two
redundant RTP streams (the original plus its duplicate) of the same
content, with each stream capable of supporting the playback when
there is no packet loss. Therefore, adding an additional RTP stream
provides a protection against packet loss. The level of protection
depends on how the packets are sent and transmitted inside the
network.
It is important to note that dual streaming can easily be extended to
support cases when more than two streams are desired. However, using
three or more streams is rare in practice, due to the high overhead
that it incurs and the little additional protection it provides.
3.1. Temporal Redundancy
From a routing perspective, two streams are considered identical if
the following two IP header fields are the same, since they will be
both routed over the same path:
o IP Source Address
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o IP Destination Address
Two routing-plane identical RTP streams might carry the same payload,
but can use different Synchronization Sources (SSRC) to differentiate
the RTP packets belonging to each stream. In the context of dual RTP
streaming, we assume that the sender duplicates the RTP packets and
sends them in separate RTP streams, each with a unique SSRC. All the
redundant streams are transmitted in the same RTP session.
For example, one main stream and its duplicate stream can be sent to
the same IP destination address and UDP destination port with a
certain delay between them [I-D.ietf-mmusic-delayed-duplication].
The streams carry the same payload in their respective RTP packets
with identical sequence numbers. This allows receivers (or other
nodes responsible for gap filling and duplicate suppression) to
identify and suppress the duplicate packets, and subsequently produce
a hopefully loss-free and duplication-free output stream. This
process is commonly called stream merging or de-duplication.
3.2. Spatial Redundancy
An RTP source might be associated with multiple network interfaces,
allowing it to send two redundant streams from two separate source
addresses. Such streams can be routed over diverse or identical
paths depending on the routing algorithm used inside the network. At
the receiving end, the node responsible for duplicate suppression can
look into various RTP header fields, for example SSRC and sequence
number, to identify and suppress the duplicate packets.
If source-specific multicast (SSM) transport is used to carry such
redundant streams, there will be a separate SSM session for each
redundant stream since the streams are sourced from different
interfaces (i.e., IP addresses). Thus, the receiving host has to
join each SSM session separately.
Alternatively, an RTP source might send the redundant streams to
separate IP destination addresses.
3.3. Dual Streaming over a Single Path or Multiple Paths
Having described the characteristics of the streams, one can reach
the following conclusions:
1. When two routing-plane identical streams are used, the two
streams will have identical IP headers. This makes it
impractical to forward the packets onto different paths. In
order to minimize packet loss, the packets belonging to one
stream are often interleaved with packets belonging to its
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duplicate stream, and with a delay, so that if there is a packet
loss, such a delay would allow the same packet from the duplicate
stream to reach the receiver because the chances that the same
packet is lost in transit again is often small. This is what is
also known as Time-shifted Redundancy, Temporal Redundancy or
simply Delayed Duplication [I-D.ietf-mmusic-delayed-duplication]
[IC2011]. This approach can be used with both types of dual
streaming, described in Section 3.1 and Section 3.2.
2. If the two streams have different IP headers, an additional
opportunity arises in that one is able to build a network, with
physically diverse paths, to deliver the two streams concurrently
to the intended receivers. This reduces the delay when packet
loss occurs and needs to be recovered. Additionally, it also
further reduces chances for packet loss. An unrecoverable loss
happens only when two network failures happen in such a way that
the same packet is affected on both paths. This is referred to
as Spatial Diversity or Spatial Redundancy [IC2011]. The
techniques used to build diverse paths are beyond the scope of
this document.
Note that spatial redundancy often offers less delay in
recovering from packet loss provided that the forwarding delay of
the network paths are more or less the same (This is often made
sure through careful network design). For both temporal and
spatial redundancy approaches, packet misordering might still
happen and needs to be handled using the sequence numbers of some
sort (e.g., RTP sequence numbers).
To summarize, dual streaming allows an application and a network to
work together to provide a near zero-loss transport with a bounded or
minimum delay. The additional advantage includes a predictable
bandwidth overhead that is proportional to the minimum bandwidth
needed for the multimedia session, but independent of the number of
receivers experiencing a packet loss and requesting a retransmission.
For a survey and comparison of similar approaches, refer to [IC2011].
3.4. Requirements
One of the following conditions is REQUIRED to hold in applications
using this specification:
o The original and duplicate RTP streams are carried (with their own
SSRCs) in the same "m" line (There could be other RTP streams
listed in the same "m" line)
o The original and duplicate RTP streams are carried in separate "m"
lines and there is no other RTP stream listed in either "m" line.
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When the original and duplicate RTP streams are carried in separate
"m" lines in an SDP description and if the SDP description has one or
more other RTP streams listed in either "m" line, duplication
grouping is not trivial and further signaling will be needed, which
is left for future standardization.
4. Use of RTP and RTCP with Temporal Redundancy
To achieve temporal redundancy, the main and duplicate RTP streams
SHOULD be sent using the sample 5-tuple of transport protocol, source
and destination IP addresses, and source and destination transport
ports. Due to the possible presence of network address and port
translation (NAPT) devices, load balancers, or other middleboxes, use
of anything other than an identical 5-tuple might also cause spatial
redundancy (which might introduce an additional delay due to the
delta between the path delays), and so is NOT RECOMMENDED unless the
path is known to be free of such middleboxes.
Since the main and duplicate RTP streams follow an identical path,
they are part of the same RTP session. Accordingly, the sender MUST
choose a different SSRC for the duplicate RTP stream than it chose
for the main RTP stream, following the rules in [RFC3550] Section 8.
4.1. RTCP Considerations
If RTCP is being sent for the main RTP stream, then the sender MUST
also generate RTCP for the duplicate RTP stream. The RTCP for the
duplicate RTP stream is generated exactly as-if the duplicate RTP
stream were a regular media stream. The sender MUST NOT duplicate
the RTCP packets sent for the main RTP stream when sending the
duplicate stream, instead it MUST generate new RTCP reports for the
duplicate stream. The sender MUST use the same RTCP CNAME in the
RTCP reports it sends for both streams, so that the receiver can
synchronize them.
The main and duplicate streams are conceptually synchronized using
the standard RTCP Sender Report-based mechanism, deriving a mapping
between their timelines. However, the RTP timestamps and sequence
numbers MUST be identical in the main and duplicate streams, making
the mapping quite trivial.
Both the main and duplicate RTP streams, and their corresponding RTCP
reports, will be received. If RTCP is used, receivers MUST generate
RTCP reports for both the main and duplicate streams in the usual
way, treating them as entirely separate media streams.
4.2. Signaling Considerations
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Signaling is needed to allow the receiver to determine that an RTP
stream is a duplicate of another, rather than a separate stream that
needs to be rendered in parallel. There are two parts to this: an
SDP extension is needed in the offer/answer exchange to negotiate
support for temporal redundancy; and signaling is needed to indicate
which stream is the duplicate (the latter can be done in-band using
an RTCP extension, or out-of-band in the SDP description).
We require out-of-band signaling for both features. The required SDP
attribute to signal duplication in the SDP offer/answer exchange
('duplication-delay') is defined in
[I-D.ietf-mmusic-delayed-duplication]. The required SDP grouping
semantics are defined in [I-D.ietf-mmusic-duplication-grouping].
In the following SDP example, a video stream is duplicated, and the
main and duplicate streams are transmitted in two separate SSRCs
(1000 and 1010):
v=0
o=ali 1122334455 1122334466 IN IP4 dup.example.com
s=Delayed Duplication
t=0 0
m=video 30000 RTP/AVP 100
c=IN IP4 233.252.0.1/127
a=source-filter:incl IN IP4 233.252.0.1 198.51.100.1
a=rtpmap:100 MP2T/90000
a=ssrc:1000 cname:ch1a@example.com
a=ssrc:1010 cname:ch1a@example.com
a=ssrc-group:DUP 1000 1010
a=duplication-delay:50
a=mid:Ch1
As specified in Section 3.2 of
[I-D.ietf-mmusic-duplication-grouping], it is advisable that the SSRC
listed first in the "a=ssrc-group:" line (i.e., SSRC of 1000) is sent
first, with the other SSRC (i.e., SSRC of 1010) being the time-
delayed duplicate. This is not critical, however, and a receiving
host should size its playout buffer based on the 'duplication-delay'
attribute, and play the stream that arrives first in preference, with
the other stream acting as a repair stream, irrespective of the order
in which they are signaled.
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5. Use of RTP and RTCP with Spatial Redundancy
When using spatial redundancy, the duplicate RTP stream is sent using
a different source and/or destination address/port pair. This will
be a separate RTP session to the session conveying the main RTP
stream. Thus, the SSRCs used for the main and duplicate streams MUST
be chosen randomly, following the rules in Section 8 of [RFC3550].
Accordingly, they will almost certainly not match each other. The
sender MUST, however, use the same RTCP CNAME for both the main and
duplicate streams. An "a=group:DUP" line or "a=ssrc-group:DUP" line
is used to indicate duplication.
5.1. RTCP Considerations
If RTCP is being sent for the main RTP stream, then the sender MUST
also generate RTCP for the duplicate RTP stream. The RTCP for the
duplicate RTP stream is generated exactly as-if the duplicate RTP
stream were a regular media stream. The sender MUST NOT duplicate
the RTCP packets sent for the main RTP stream when sending the
duplicate stream, instead it MUST generate new RTCP reports for the
duplicate stream. The sender MUST use the same RTCP CNAME in the
RTCP reports it sends for both streams, so that the receiver can
synchronize them.
The main and duplicate streams are conceptually synchronized using
the standard RTCP Sender Report-based mechanism, deriving a mapping
between their timelines. However, the RTP timestamps and sequence
numbers MUST be identical in the main and duplicate streams, making
the mapping quite trivial.
Both the main and duplicate RTP streams, and their corresponding RTCP
reports, will be received. If RTCP is used, receivers MUST generate
RTCP reports for both the main and duplicate streams in the usual
way, treating them as entirely separate media streams.
5.2. Signaling Considerations
The required SDP grouping semantics have been defined in
[I-D.ietf-mmusic-duplication-grouping]. In the following example,
the redundant streams have different IP destination addresses. The
example shows the same UDP port number and IP source address for each
stream, but either or both could have been different for the two
streams.
v=0
o=ali 1122334455 1122334466 IN IP4 dup.example.com
s=DUP Grouping Semantics
t=0 0
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a=group:DUP S1a S1b
m=video 30000 RTP/AVP 100
c=IN IP4 233.252.0.1/127
a=source-filter:incl IN IP4 233.252.0.1 198.51.100.1
a=rtpmap:100 MP2T/90000
a=mid:S1a
m=video 30000 RTP/AVP 101
c=IN IP4 233.252.0.2/127
a=source-filter:incl IN IP4 233.252.0.2 198.51.100.1
a=rtpmap:101 MP2T/90000
a=mid:S1b
6. Use of RTP and RTCP with Temporal and Spatial Redundancy
This uses the same RTP/RTCP mechanisms from Sections Section 4 and
Section 5, plus a combination of both sets of signaling.
7. Congestion Control Considerations
Duplicating RTP streams has several considerations in the context of
congestion control. First of all, RTP duplication MUST NOT be used
in cases where the primary cause of packet loss is congestion since
duplication can make congestion only worse. Furthermore, RTP
duplication SHOULD NOT be used where there is a risk of congestion
upon duplicating an RTP stream. Duplication is RECOMMENDED only to
be used for protection against network outages due to a temporary
link or network element failure and where it is known that there is
sufficient network capacity to carry the duplicated traffic. The
capacity requirement constrains the use of duplication to managed
networks, and makes it unsuitable for use on unmanaged public
networks.
It is essential that the nodes responsible for the duplication and
de-duplication are aware of the original stream's requirements and
the available capacity inside the network. If there is an adaptation
capability for the original stream, these nodes have to assume the
same adaptation capability for the duplicated stream, too. For
example, if the source doubles the bitrate for the original stream,
the bitrate of the duplicate stream will also be doubled.
Depending on where de-duplication takes place, there could be
different scenarios. When the duplication and de-duplication takes
place inside the network before the ultimate end-points that will
consume the RTP media, the whole process is transparent to these end-
points. Thus, these end-points will apply any congestion control, if
applicable, on the de-duplicated RTP stream. This output stream will
have less losses than either of the original and duplicated stream,
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and the end-point will make congestion control decisions accordingly.
However, if de-duplication takes place at the ultimate end-point,
this end-point MUST consider the aggregate of the original and
duplicated RTP stream in any congestion control it wants to apply.
The end-point will observe the losses in each stream separately, and
this information can be used to fine-tune the duplication process.
For example, the duplication interval can be adjusted based on the
duration of a common packet loss in both streams.
8. Security Considerations
The security considerations of [RFC3550],
[I-D.ietf-mmusic-delayed-duplication],
[I-D.ietf-mmusic-duplication-grouping], and any RTP profiles and
payload formats in use apply.
Duplication can be performed end-to-end, with the media sender
generating a duplicate RTP stream, and the receiver(s) performing de-
duplication. In such cases, if the original media stream is to be
authenticated (e.g., using SRTP [RFC3711]) then the duplicate stream
also needs to be authenticated, and duplicate packets that fail the
authentication check need to be discarded.
Stream duplication and de-duplication can also be performed by in-
network middleboxes. Such middleboxes will need to rewrite the RTP
SSRC such that the RTP packets in the duplicate stream have a
different SSRC to the original stream, and will need to generate and
respond to RTCP packets corresponding to the duplicate stream. This
sort of in-network duplication service has the potential to act as an
amplifier for denial-of-service attacks if the attacker can cause
attack traffic to be duplicated. To prevent this, middleboxes
providing the duplication service need to authenticate the traffic to
be duplicated as being from a legitimate source, for example using
the secure RTP (SRTP) profile [RFC3711]. This requires the middlebox
to be part of the security context of the media session being
duplicated, so it has access to the necessary keying material for
authentication. To do this, the middlebox will need to be privy to
the session set-up signalling. Details of how that is done will
depend on the type of signalling used (SIP, RTSP, WebRTC, etc.), and
is not specified here.
Similarly, to prevent packet injection attacks, a de-duplication
middlebox needs to authenticate original and duplicate streams, and
ought not use non-authenticated packets that are received. Again,
this requires the middlebox to be part of the security context, and
have access to the appropriate signalling and keying material.
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The use of the encryption features of SRTP does not affect stream de-
duplication middleboxes, since the RTP headers are sent in the clear.
9. IANA Considerations
No IANA actions are required.
10. Acknowledgments
Thanks to Magnus Westerlund for his suggestions.
11. References
11.1. Normative References
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[I-D.ietf-mmusic-delayed-duplication]
Begen, A., Cai, Y., and H. Ou, "Delayed Duplication
Attribute in the Session Description Protocol", draft-
ietf-mmusic-delayed-duplication-02 (work in progress), May
2013.
[I-D.ietf-mmusic-duplication-grouping]
Begen, A., Cai, Y., and H. Ou, "Duplication Grouping
Semantics in the Session Description Protocol", draft-
ietf-mmusic-duplication-grouping-03 (work in progress),
July 2013.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
11.2. Informative References
[RFC2354] Perkins, C. and O. Hodson, "Options for Repair of
Streaming Media", RFC 2354, June 1998.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
July 2006.
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[RFC6363] Watson, M., Begen, A., and V. Roca, "Forward Error
Correction (FEC) Framework", RFC 6363, October 2011.
[IC2011] Evans, J., Begen, A., Greengrass, J., and C. Filsfils,
"Toward Lossless Video Transport (to appear in IEEE
Internet Computing)", November 2011.
Authors' Addresses
Ali Begen
Cisco
181 Bay Street
Toronto, ON M5J 2T3
CANADA
Email: abegen@cisco.com
Colin Perkins
University of Glasgow
School of Computing Science
Glasgow G12 8QQ
UK
Email: csp@csperkins.org
URI: http://orcid.org/0000-0002-3404-8964
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