draft-ietf-avtext-rtp-duplication-01.txt   draft-ietf-avtext-rtp-duplication-02.txt 
AVTEXT A. Begen AVTEXT A. Begen
Internet-Draft Cisco Internet-Draft Cisco
Intended status: Standards Track C. Perkins Intended status: Standards Track C. Perkins
Expires: July 3, 2013 University of Glasgow Expires: September 22, 2013 University of Glasgow
December 30, 2012 March 21, 2013
Duplicating RTP Streams Duplicating RTP Streams
draft-ietf-avtext-rtp-duplication-01 draft-ietf-avtext-rtp-duplication-02
Abstract Abstract
Packet loss is undesirable for real-time multimedia sessions, but can Packet loss is undesirable for real-time multimedia sessions, but can
occur due to congestion, or other unplanned network outages. This is occur due to congestion, or other unplanned network outages. This is
especially true for IP multicast networks, where packet loss patterns especially true for IP multicast networks, where packet loss patterns
can vary greatly between receivers. One technique that can be used can vary greatly between receivers. One technique that can be used
to recover from packet loss without incurring unbounded delay for all to recover from packet loss without incurring unbounded delay for all
the receivers is to duplicate the packets and send them in separate the receivers is to duplicate the packets and send them in separate
redundant streams. This document explains how Real-time Transport redundant streams. This document explains how Real-time Transport
Protocol (RTP) streams can be duplicated without breaking RTP media Protocol (RTP) streams can be duplicated without breaking RTP or RTP
streams, or RTP Control Protocol (RTCP) rules. Control Protocol (RTCP) rules.
Status of this Memo Status of This Memo
This Internet-Draft is submitted in full conformance with the This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/. Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on July 3, 2013. This Internet-Draft will expire on September 22, 2013.
Copyright Notice Copyright Notice
Copyright (c) 2012 IETF Trust and the persons identified as the Copyright (c) 2013 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of (http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License. described in the Simplified BSD License.
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Terminology and Requirements Notation . . . . . . . . . . . . 3 2. Terminology and Requirements Notation . . . . . . . . . . . . 3
3. Dual Streaming Use Cases . . . . . . . . . . . . . . . . . . . 3 3. Dual Streaming Use Cases . . . . . . . . . . . . . . . . . . 3
3.1. Temporal Redundancy . . . . . . . . . . . . . . . . . . . 4 3.1. Temporal Redundancy . . . . . . . . . . . . . . . . . . . 3
3.2. Spatial Redundancy . . . . . . . . . . . . . . . . . . . . 4 3.2. Spatial Redundancy . . . . . . . . . . . . . . . . . . . 4
3.3. Dual Streaming over a Single Path or Multiple Paths . . . 5 3.3. Dual Streaming over a Single Path or Multiple Paths . . . 4
4. Use of RTP and RTCP with Temporal Redundancy . . . . . . . . . 6 4. Use of RTP and RTCP with Temporal Redundancy . . . . . . . . 5
4.1. RTCP Considerations . . . . . . . . . . . . . . . . . . . 6 4.1. RTCP Considerations . . . . . . . . . . . . . . . . . . . 6
4.2. Signaling Considerations . . . . . . . . . . . . . . . . . 6 4.2. Signaling Considerations . . . . . . . . . . . . . . . . 6
5. Use of RTP and RTCP with Spatial Redundancy . . . . . . . . . 7 5. Use of RTP and RTCP with Spatial Redundancy . . . . . . . . . 7
5.1. RTCP Considerations . . . . . . . . . . . . . . . . . . . 8 5.1. RTCP Considerations . . . . . . . . . . . . . . . . . . . 7
5.2. Signaling Considerations . . . . . . . . . . . . . . . . . 8 5.2. Signaling Considerations . . . . . . . . . . . . . . . . 8
6. Use of RTP and RTCP with Temporal and Spatial Redundancy . . . 9 6. Use of RTP and RTCP with Temporal and Spatial Redundancy . . 8
7. Security Considerations . . . . . . . . . . . . . . . . . . . 9 7. Security Considerations . . . . . . . . . . . . . . . . . . . 8
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9
9. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 10 9. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 9
10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 10 10. References . . . . . . . . . . . . . . . . . . . . . . . . . 9
10.1. Normative References . . . . . . . . . . . . . . . . . . . 10 10.1. Normative References . . . . . . . . . . . . . . . . . . 9
10.2. Informative References . . . . . . . . . . . . . . . . . . 10 10.2. Informative References . . . . . . . . . . . . . . . . . 9
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 11 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 10
1. Introduction 1. Introduction
The Real-time Transport Protocol (RTP) [RFC3550] is widely used today The Real-time Transport Protocol (RTP) [RFC3550] is widely used today
for delivering IPTV traffic, and other real-time multimedia sessions. for delivering IPTV traffic, and other real-time multimedia sessions.
Many of these applications support very large numbers of receivers, Many of these applications support very large numbers of receivers,
and rely on intra-domain UDP/IP multicast for efficient distribution and rely on intra-domain UDP/IP multicast for efficient distribution
of traffic within the network. of traffic within the network.
While this combination has proved successful, there does exist a While this combination has proved successful, there does exist a
skipping to change at page 3, line 33 skipping to change at page 3, line 15
One technique to recover from packet loss without incurring unbounded One technique to recover from packet loss without incurring unbounded
delay for all the receivers is to duplicate the packets and send them delay for all the receivers is to duplicate the packets and send them
in separate redundant streams. Variations on this idea have been in separate redundant streams. Variations on this idea have been
implemented and deployed today [IC2011]. However, duplication of RTP implemented and deployed today [IC2011]. However, duplication of RTP
streams without breaking the RTP and RTCP functionality has not been streams without breaking the RTP and RTCP functionality has not been
documented properly. This document explains how duplication can be documented properly. This document explains how duplication can be
achieved for RTP streams. achieved for RTP streams.
Stream duplication offers a simple way to protect media flows from Stream duplication offers a simple way to protect media flows from
packet loss. It has a comparatively high bandwidth overhead, since packet loss. It has a comparatively high bandwidth overhead, since
everything is sent twice, but with a low processor overhead. It is everything is sent twice, but with a low processing overhead. It is
also very predictable in its overheads. Alternative approaches may also very predictable in its overheads. Alternative approaches may
be suitable in some cases, for example retransmission-based recovery be suitable in some cases, for example retransmission-based recovery
[RFC4588] or forward error correction [RFC5109]. [RFC4588] or Forward Error Correction [RFC6363].
2. Terminology and Requirements Notation 2. Terminology and Requirements Notation
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in "OPTIONAL" in this document are to be interpreted as described in
[RFC2119]. [RFC2119].
3. Dual Streaming Use Cases 3. Dual Streaming Use Cases
Dual streaming refers to a technique that involves transmitting two Dual streaming refers to a technique that involves transmitting two
redundant RTP streams of the same content, with each stream capable redundant RTP streams (the original plus its duplicate) of the same
of supporting the playback when there is no packet loss. Therefore, content, with each stream capable of supporting the playback when
adding an additional RTP stream provides a protection against packet there is no packet loss. Therefore, adding an additional RTP stream
loss. The level of protection depends on how the packets are sent provides a protection against packet loss. The level of protection
and transmitted inside the network. 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 It is important to note that dual streaming can easily be extended to
support cases when more than two streams are desired. However, using support cases when more than two streams are desired. However, using
three or more streams is rare in practise, due to the high overhead three or more streams is rare in practice, due to the high overhead
that it incurs. that it incurs.
3.1. Temporal Redundancy 3.1. Temporal Redundancy
From a routing perspective, two streams are considered identical if From a routing perspective, two streams are considered identical if
the following two IP header fields are the same, since they will be the following two IP header fields are the same, since they will be
both routed over the same path: both routed over the same path:
o IP Source Address o IP Source Address
skipping to change at page 4, line 21 skipping to change at page 4, line 4
3.1. Temporal Redundancy 3.1. Temporal Redundancy
From a routing perspective, two streams are considered identical if From a routing perspective, two streams are considered identical if
the following two IP header fields are the same, since they will be the following two IP header fields are the same, since they will be
both routed over the same path: both routed over the same path:
o IP Source Address o IP Source Address
o IP Destination Address o IP Destination Address
Two routing-plane identical RTP streams might carry the same payload, Two routing-plane identical RTP streams might carry the same payload,
but can use different Synchronization Sources (SSRC) to differentiate but can use different Synchronization Sources (SSRC) to differentiate
the RTP packets belonging to each stream. In the context of dual RTP the RTP packets belonging to each stream. In the context of dual RTP
streaming, we assume that the source duplicates the RTP packets and streaming, we assume that the sender duplicates the RTP packets and
sends them in separate RTP streams, each with a unique SSRC. All the sends them in separate RTP streams, each with a unique SSRC. All the
redundant streams are transmitted in the same RTP session. redundant streams are transmitted in the same RTP session.
For example, one main and one redundant RTP stream can be sent to the For example, one main stream and its duplicate stream can be sent to
same IP destination address and UDP destination port with a certain the same IP destination address and UDP destination port with a
delay between them [I-D.ietf-mmusic-delayed-duplication]. The certain delay between them [I-D.ietf-mmusic-delayed-duplication].
streams carry the same payload in their respective RTP packets with The streams carry the same payload in their respective RTP packets
identical sequence numbers. This allows receivers (or other nodes with identical sequence numbers. This allows receivers (or other
responsible for gap filling and duplicate suppression) to identify nodes responsible for gap filling and duplicate suppression) to
and suppress the duplicate packets, and subsequently produce a identify and suppress the duplicate packets, and subsequently produce
hopefully loss-free and duplication-free output stream. This process a hopefully loss-free and duplication-free output stream. This
is called stream merging. process is commonly called stream merging or de-duplication.
3.2. Spatial Redundancy 3.2. Spatial Redundancy
An RTP source might be associated with multiple network interfaces, An RTP source might be associated with multiple network interfaces,
allowing it to send two redundant streams from two separate source allowing it to send two redundant streams from two separate source
addresses. Such streams can be routed over diverse or identical addresses. Such streams can be routed over diverse or identical
paths depending on the routing algorithm used inside the network. At paths depending on the routing algorithm used inside the network. At
the receiving end, the node responsible for duplicate suppression can the receiving end, the node responsible for duplicate suppression can
look into various RTP header fields, for example SSRC and sequence look into various RTP header fields, for example SSRC and sequence
number, to identify and suppress the duplicate packets. number, to identify and suppress the duplicate packets.
skipping to change at page 5, line 19 skipping to change at page 4, line 49
3.3. Dual Streaming over a Single Path or Multiple Paths 3.3. Dual Streaming over a Single Path or Multiple Paths
Having described the characteristics of the streams, one can reach Having described the characteristics of the streams, one can reach
the following conclusions: the following conclusions:
1. When two routing-plane identical streams are used, the two 1. When two routing-plane identical streams are used, the two
streams will have identical IP headers. This makes it streams will have identical IP headers. This makes it
impractical to forward the packets onto different paths. In impractical to forward the packets onto different paths. In
order to minimize packet loss, the packets belonging to one order to minimize packet loss, the packets belonging to one
stream are often interleaved with packets belonging to the other, stream are often interleaved with packets belonging to its
and with a delay, so that if there is a packet loss, such a delay duplicate stream, and with a delay, so that if there is a packet
would allow the same packet from the other stream to reach the loss, such a delay would allow the same packet from the duplicate
receiver because the chances that the same packet is lost in stream to reach the receiver because the chances that the same
transit again is often small. This is what is also known as packet is lost in transit again is often small. This is what is
Time-shifted Redundancy, Temporal Redundancy or simply Delayed also known as Time-shifted Redundancy, Temporal Redundancy or
Duplication [I-D.ietf-mmusic-delayed-duplication] [IC2011]. This simply Delayed Duplication [I-D.ietf-mmusic-delayed-duplication]
approach can be used with both types of dual streaming, described [IC2011]. This approach can be used with both types of dual
in Section 3.1 and Section 3.2. streaming, described in Section 3.1 and Section 3.2.
2. If the two streams have different IP headers, an additional 2. If the two streams have different IP headers, an additional
opportunity arises in that one is able to build a network, with opportunity arises in that one is able to build a network, with
physically diverse paths, to deliver the two streams concurrently physically diverse paths, to deliver the two streams concurrently
to the intended receivers. This reduces the delay when packet to the intended receivers. This reduces the delay when packet
loss occurs and needs to be recovered. Additionally, it also loss occurs and needs to be recovered. Additionally, it also
further reduces chances for packet loss. An unrecoverable loss further reduces chances for packet loss. An unrecoverable loss
happens only when two network failures happen in such a way that happens only when two network failures happen in such a way that
the same packet is affected on both paths. This is referred to the same packet is affected on both paths. This is referred to
as Spatial Diversity or Spatial Redundancy [IC2011]. The as Spatial Diversity or Spatial Redundancy [IC2011]. The
techniques used to build diverse paths are beyond the scope of techniques used to build diverse paths are beyond the scope of
this document. this document.
Note that spatial redundancy often offers less delay in Note that spatial redundancy often offers less delay in
recovering from packet loss provided that the forwarding delay of recovering from packet loss provided that the forwarding delay of
the network paths are more or less the same. For both temporal the network paths are more or less the same (This is often made
and spatial redundancy approaches, packet misordering might still 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 happen and needs to be handled using the sequence numbers of some
sort (e.g., RTP sequence numbers). sort (e.g., RTP sequence numbers).
To summarize, dual streaming allows an application and a network to To summarize, dual streaming allows an application and a network to
work together to provide a near zero-loss transport with a bounded or work together to provide a near zero-loss transport with a bounded or
minimum delay. The additional advantage includes a predictable minimum delay. The additional advantage includes a predictable
bandwidth overhead that is proportional to the minimum bandwidth bandwidth overhead that is proportional to the minimum bandwidth
needed for the multimedia session, but independent of the number of needed for the multimedia session, but independent of the number of
receivers experiencing a packet loss and requesting a retransmission. receivers experiencing a packet loss and requesting a retransmission.
For a survey and comparison of similar approaches, refer to [IC2011]. For a survey and comparison of similar approaches, refer to [IC2011].
4. Use of RTP and RTCP with Temporal Redundancy 4. Use of RTP and RTCP with Temporal Redundancy
To achieve temporal redundancy, the main and redundant RTP streams To achieve temporal redundancy, the main and duplicate RTP streams
MUST be sent using the same 5-tuple of transport protocol, source and SHOULD be sent using the sample 5-tuple of transport protocol, source
destination IP addresses, and source and destination transport ports. and destination IP addresses, and source and destination transport
This is perhaps overly restrictive, but with the possible presence of ports. Due to the possible presence of network address and port
network address and port translation (NAPT) devices, using anything translation (NAPT) devices, load balancers, or other middleboxes, use
other than an identical 5-tuple can also cause spatial redundancy. 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 main and redundant RTP streams follow an identical path, they Since the main and duplicate RTP streams follow an identical path,
are part of the same RTP session. Accordingly, the sender MUST they are part of the same RTP session. Accordingly, the sender MUST
choose a different SSRC for the redundant RTP stream than it chose choose a different SSRC for the duplicate RTP stream than it chose
for the main RTP stream, following the rules in [RFC3550] Section 8. for the main RTP stream, following the rules in [RFC3550] Section 8.
4.1. RTCP Considerations 4.1. RTCP Considerations
If RTCP is being sent for the main RTP stream, then the sender MUST If RTCP is being sent for the main RTP stream, then the sender MUST
also generate RTCP for the redundant RTP stream. The RTCP for the also generate RTCP for the duplicate RTP stream. The RTCP for the
redundant RTP stream is generated exactly as-if the redundant RTP duplicate RTP stream is generated exactly as-if the duplicate RTP
stream were a regular media stream. The sender MUST NOT duplicate stream were a regular media stream. The sender MUST NOT duplicate
the RTCP packets sent for the main RTP stream when sending the 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, instead it MUST generate new RTCP reports for the
duplicate stream. The sender MUST use the same RTCP CNAME in the duplicate stream. The sender MUST use the same RTCP CNAME in the
RTCP reports it sends for the main and redundant streams, so that the RTCP reports it sends for both streams, so that the receiver can
receiver can synchronize them. synchronize them.
Both the main and redundant RTP streams, and their corresponding RTCP 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 reports, will be received. If RTCP is used, receivers MUST generate
RTCP reports for both main and redundant streams in the usual way, RTCP reports for both the main and duplicate streams in the usual
treating them as entirely separate media streams. way, treating them as entirely separate media streams.
4.2. Signaling Considerations 4.2. Signaling Considerations
Signaling is needed to allow the receiver to determine that an RTP Signaling is needed to allow the receiver to determine that an RTP
stream is a redundant copy of another, rather than a separate stream stream is a duplicate of another, rather than a separate stream that
that needs to be rendered in parallel. There are two parts to this: needs to be rendered in parallel. There are two parts to this: an
an SDP extension is needed in the offer/answer exchange to negotiate SDP extension is needed in the offer/answer exchange to negotiate
support for temporal redundancy; and signaling is needed to indicate support for temporal redundancy; and signaling is needed to indicate
which stream is the duplicate (the latter can be done in-band using which stream is the duplicate (the latter can be done in-band using
an RTCP extension, or out-of-band by signaling the SSRCs used by the an RTCP extension, or out-of-band in the SDP description).
duplicate streams in SDP).
We require out-of-band signaling for both features. The required SDP We require out-of-band signaling for both features. The required SDP
attribute to signal duplication in the SDP offer/answer exchange attribute to signal duplication in the SDP offer/answer exchange
('duplication-delay') is defined in ('duplication-delay') is defined in
[I-D.ietf-mmusic-delayed-duplication]. The required SDP grouping [I-D.ietf-mmusic-delayed-duplication]. The required SDP grouping
semantics are defined in [I-D.ietf-mmusic-duplication-grouping]. semantics are defined in [I-D.ietf-mmusic-duplication-grouping].
In the following SDP example, a video stream is duplicated, and the In the following SDP example, a video stream is duplicated, and the
main and redundant streams are transmitted in two separate SSRCs main and duplicate streams are transmitted in two separate SSRCs
(1000 and 1010): (1000 and 1010):
v=0 v=0
o=ali 1122334455 1122334466 IN IP4 dup.example.com o=ali 1122334455 1122334466 IN IP4 dup.example.com
s=Delayed Duplication s=Delayed Duplication
t=0 0 t=0 0
m=video 30000 RTP/AVP 100 m=video 30000 RTP/AVP 100
c=IN IP4 233.252.0.1/127 c=IN IP4 233.252.0.1/127
a=source-filter:incl IN IP4 233.252.0.1 198.51.100.1 a=source-filter:incl IN IP4 233.252.0.1 198.51.100.1
a=rtpmap:100 MP2T/90000 a=rtpmap:100 MP2T/90000
a=ssrc:1000 cname:ch1@example.com a=ssrc:1000 cname:ch1a@example.com
a=ssrc:1010 cname:ch1@example.com a=ssrc:1010 cname:ch1a@example.com
a=ssrc-group:DUP 1000 1010 a=ssrc-group:DUP 1000 1010
a=duplication-delay:100 a=duplication-delay:50
a=mid:Group1 a=mid:Ch1
It is RECOMMENDED that the SSRC listed first in the "a=ssrc-group:" As specified in Section 3.2 of
line is sent first, with the other RTP SSRC being the time-delayed [I-D.ietf-mmusic-duplication-grouping], it is advisable that the SSRC
duplicate. This is not critical, however, and receivers should size listed first in the "a=ssrc-group:" line (i.e., SSRC of 1000) is sent
their playout buffers based on the "a=duplication-delay:" attribute, first, with the other SSRC (i.e., SSRC of 1010) being the time-
and play the stream that arrives first in preference, with the other delayed duplicate. This is not critical, however, and a receiving
stream acting as a repair stream, irrespective of the order in which host should size its playout buffer based on the 'duplication-delay'
they are signalled. 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.
5. Use of RTP and RTCP with Spatial Redundancy 5. Use of RTP and RTCP with Spatial Redundancy
When using spatial redundancy, the redundant RTP stream is sent on When using spatial redundancy, the duplicate RTP stream is sent using
using a different source and/or destination address/port pair. This a different source and/or destination address/port pair. This will
will be a separate RTP session to the session conveying the main RTP be a separate RTP session to the session conveying the main RTP
stream. stream. Thus, the SSRCs used for the main and duplicate streams MUST
be chosen randomly, following the rules in Section 8 of [RFC3550].
The SSRCs used for the main and redundant streams MUST be chosen
randomly, following the rules in Section 8 of [RFC3550].
Accordingly, they will almost certainly not match each other. The Accordingly, they will almost certainly not match each other. The
sender MUST, however, use the same RTCP CNAME for both the main and sender MUST, however, use the same RTCP CNAME for both the main and
redundant streams, and MUST include an "a=ssrc:... srcname:..." duplicate streams. An "a=group:DUP" line or "a=ssrc-group:DUP" line
attribute to correlate the flows. An "a=group:DUP" attribute is used is used to indicate duplication.
to indicate duplication.
5.1. RTCP Considerations 5.1. RTCP Considerations
If RTCP is being sent for the main RTP stream, then the sender MUST If RTCP is being sent for the main RTP stream, then the sender MUST
also generate RTCP for the redundant RTP stream. The RTCP for the also generate RTCP for the duplicate RTP stream. The RTCP for the
redundant RTP stream is generated exactly as-if the redundant RTP duplicate RTP stream is generated exactly as-if the duplicate RTP
stream were a regular media stream; the sender MUST NOT duplicate the stream were a regular media stream. The sender MUST NOT duplicate
RTCP packets sent for the main RTP stream. The sender MUST use the the RTCP packets sent for the main RTP stream when sending the
same RTCP CNAME in the RTCP reports it sends for the main and duplicate stream, instead it MUST generate new RTCP reports for the
redundant streams, so that the receiver can synchronize them. 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 redundant streams are conceptually synchronised using The main and duplicate streams are conceptually synchronized using
the standard RTCP SR-based mechanism, deriving a mapping between the standard RTCP Sender Report-based mechanism, deriving a mapping
their timelines. The RTP timestamps and sequence numbers SHOULD be between their timelines. However, the RTP timestamps and sequence
identical in the main and redundant streams, however, making the numbers MUST be identical in the main and duplicate streams, making
mapping trivial in most cases. the mapping quite trivial.
Both main and redundant streams, and their corresponding RTCP, will Both the main and duplicate RTP streams, and their corresponding RTCP
be received. If RTCP is used, receivers MUST generate RTCP reports reports, will be received. If RTCP is used, receivers MUST generate
for both main and redundant streams in the usual way, treating them RTCP reports for both the main and duplicate streams in the usual
as entirely separate media streams. way, treating them as entirely separate media streams.
5.2. Signaling Considerations 5.2. Signaling Considerations
The required SDP grouping semantics have been defined in The required SDP grouping semantics have been defined in
[I-D.ietf-mmusic-duplication-grouping]. In the following example, [I-D.ietf-mmusic-duplication-grouping]. In the following example,
the redundant streams have different IP destination addresses. The the redundant streams have different IP destination addresses. The
example shows the same UDP port number and IP source addresses, but example shows the same UDP port number and IP source address for each
either or both could have been different for the two streams. stream, but either or both could have been different for the two
streams.
v=0 v=0
o=ali 1122334455 1122334466 IN IP4 dup.example.com o=ali 1122334455 1122334466 IN IP4 dup.example.com
s=DUP Grouping Semantics s=DUP Grouping Semantics
t=0 0 t=0 0
a=group:DUP S1a S1b a=group:DUP S1a S1b
m=video 30000 RTP/AVP 100 m=video 30000 RTP/AVP 100
c=IN IP4 233.252.0.1/127 c=IN IP4 233.252.0.1/127
a=source-filter:incl IN IP4 233.252.0.1 198.51.100.1 a=source-filter:incl IN IP4 233.252.0.1 198.51.100.1
a=rtpmap:100 MP2T/90000 a=rtpmap:100 MP2T/90000
a=ssrc:1000 cname:ch1@example.com
a=ssrc:1000 srcname:45:a8:f4:19:b4:c3
a=mid:S1a a=mid:S1a
m=video 30000 RTP/AVP 101 m=video 30000 RTP/AVP 101
c=IN IP4 233.252.0.2/127 c=IN IP4 233.252.0.2/127
a=source-filter:incl IN IP4 233.252.0.2 198.51.100.1 a=source-filter:incl IN IP4 233.252.0.2 198.51.100.1
a=rtpmap:101 MP2T/90000 a=rtpmap:101 MP2T/90000
a=ssrc:1010 cname:ch1@example.com
a=ssrc:1010 srcname:45:a8:f4:19:b4:c3
a=mid:S1b a=mid:S1b
6. Use of RTP and RTCP with Temporal and Spatial Redundancy 6. Use of RTP and RTCP with Temporal and Spatial Redundancy
This uses the same RTP/RTCP mechanisms, plus a combination of both This uses the same RTP/RTCP mechanisms from Sections Section 4 and
sets of signaling. Section 5, plus a combination of both sets of signaling.
7. Security Considerations 7. Security Considerations
The security considerations of [RFC3550], The security considerations of [RFC3550],
[I-D.ietf-mmusic-delayed-duplication], and [I-D.ietf-mmusic-delayed-duplication], and
[I-D.ietf-mmusic-duplication-grouping] apply. [I-D.ietf-mmusic-duplication-grouping] apply.
If stream de-duplication is done by an in-network middlebox, rather Stream de-duplication can be done by an in-network middlebox,
than by an end system, that middlebox can work if Secure RTP (SRTP) rewriting the SSRC as appropriate. If the Secure RTP (SRTP) profile
encryption is used [RFC3711], since the RTP headers are in the clear. [RFC3711] is used to authenticate RTP packets, such rewriting is not
Doing so would break the authentication when the SSRC is rewritten, possible without breaking the authentication, unless the de-
unless the de-duplication middlebox were trusted to re-authenticate duplication middlebox is trusted to re-authenticate the packets.
the packets. This would require additional signaling which is not This would require additional signaling that is not specified here.
specified here, since de-duplication in the receiver end system is The use of the encryption features of SRTP does not affect stream de-
expected to be the more common use case. duplication middleboxes, since the RTP headers are sent in the clear.
8. IANA Considerations 8. IANA Considerations
No IANA actions are required. No IANA actions are required.
9. Acknowledgments 9. Acknowledgments
Thanks to Magnus Westerlund for his suggestions. Thanks to Magnus Westerlund for his suggestions.
10. References 10. References
skipping to change at page 10, line 22 skipping to change at page 9, line 35
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003. Applications", STD 64, RFC 3550, July 2003.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997. Requirement Levels", BCP 14, RFC 2119, March 1997.
[I-D.ietf-mmusic-delayed-duplication] [I-D.ietf-mmusic-delayed-duplication]
Begen, A., Cai, Y., and H. Ou, "Delayed Duplication Begen, A., Cai, Y., and H. Ou, "Delayed Duplication
Attribute in the Session Description Protocol", Attribute in the Session Description Protocol", draft-
draft-ietf-mmusic-delayed-duplication-00 (work in ietf-mmusic-delayed-duplication-01 (work in progress),
progress), October 2012. March 2013.
[I-D.ietf-mmusic-duplication-grouping] [I-D.ietf-mmusic-duplication-grouping]
Begen, A., Cai, Y., and H. Ou, "Duplication Grouping Begen, A., Cai, Y., and H. Ou, "Duplication Grouping
Semantics in the Session Description Protocol", Semantics in the Session Description Protocol", draft-
draft-ietf-mmusic-duplication-grouping-00 (work in ietf-mmusic-duplication-grouping-01 (work in progress),
progress), October 2012. March 2013.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)", Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004. RFC 3711, March 2004.
10.2. Informative References 10.2. Informative References
[RFC2354] Perkins, C. and O. Hodson, "Options for Repair of [RFC2354] Perkins, C. and O. Hodson, "Options for Repair of
Streaming Media", RFC 2354, June 1998. Streaming Media", RFC 2354, June 1998.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R. [RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588, Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
July 2006. July 2006.
[RFC5109] Li, A., "RTP Payload Format for Generic Forward Error [RFC6363] Watson, M., Begen, A., and V. Roca, "Forward Error
Correction", RFC 5109, December 2007. Correction (FEC) Framework", RFC 6363, October 2011.
[IC2011] Evans, J., Begen, A., Greengrass, J., and C. Filsfils, [IC2011] Evans, J., Begen, A., Greengrass, J., and C. Filsfils,
"Toward Lossless Video Transport (to appear in IEEE "Toward Lossless Video Transport (to appear in IEEE
Internet Computing)", November 2011. Internet Computing)", November 2011.
Authors' Addresses Authors' Addresses
Ali Begen Ali Begen
Cisco Cisco
181 Bay Street 181 Bay Street
Toronto, ON M5J 2T3 Toronto, ON M5J 2T3
CANADA CANADA
Email: abegen@cisco.com Email: abegen@cisco.com
Colin Perkins Colin Perkins
University of Glasgow University of Glasgow
School of Computing Science School of Computing Science
Glasgow, G12 8QQ Glasgow G12 8QQ
UK UK
Email: csp@csperkins.org Email: csp@csperkins.org
 End of changes. 46 change blocks. 
137 lines changed or deleted 144 lines changed or added

This html diff was produced by rfcdiff 1.33. The latest version is available from http://tools.ietf.org/tools/rfcdiff/