path: root/doc/rfc/rfc7198.txt

Internet Engineering Task Force (IETF)                          A. Begen
Request for Comments: 7198                                         Cisco
Category: Standards Track                                     C. Perkins
ISSN: 2070-1721                                    University of Glasgow
                                                              April 2014


                        Duplicating RTP Streams

Abstract

   Packet loss is undesirable for real-time multimedia sessions but can
   occur due to a variety of reasons including unplanned network
   outages.  In unicast transmissions, recovering from such an outage
   can be difficult depending on the outage duration, due to the
   potentially large number of missing packets.  In multicast
   transmissions, recovery is even more challenging as many receivers
   could be impacted by the outage.  For this challenge, one solution
   that does not incur unbounded delay is to duplicate the packets and
   send them in separate redundant streams, provided that the underlying
   network satisfies certain requirements.  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

   This is an Internet Standards Track document.

   This document is a product of the Internet Engineering Task Force
   (IETF).  It represents the consensus of the IETF community.  It has
   received public review and has been approved for publication by the
   Internet Engineering Steering Group (IESG).  Further information on
   Internet Standards is available in Section 2 of RFC 5741.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at
   http://www.rfc-editor.org/info/rfc7198.














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Copyright Notice

   Copyright (c) 2014 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Terminology and Requirements Notation . . . . . . . . . . . .   4
   3.  Use Cases for Dual Streaming  . . . . . . . . . . . . . . . .   4
     3.1.  Temporal Redundancy . . . . . . . . . . . . . . . . . . .   4
     3.2.  Spatial Redundancy  . . . . . . . . . . . . . . . . . . .   5
     3.3.  Dual Streaming over a Single Path or Multiple Paths . . .   5
     3.4.  Requirements  . . . . . . . . . . . . . . . . . . . . . .   6
   4.  Use of RTP and RTCP with Temporal Redundancy  . . . . . . . .   7
     4.1.  RTCP Considerations . . . . . . . . . . . . . . . . . . .   7
     4.2.  Signaling Considerations  . . . . . . . . . . . . . . . .   7
   5.  Use of RTP and RTCP with Spatial Redundancy . . . . . . . . .   8
     5.1.  RTCP Considerations . . . . . . . . . . . . . . . . . . .   9
     5.2.  Signaling Considerations  . . . . . . . . . . . . . . . .   9
   6.  Use of RTP and RTCP with Temporal and Spatial Redundancy  . .  10
   7.  Congestion Control Considerations . . . . . . . . . . . . . .  10
   8.  Security Considerations . . . . . . . . . . . . . . . . . . .  11
   9.  Acknowledgments . . . . . . . . . . . . . . . . . . . . . . .  11
   10. References  . . . . . . . . . . . . . . . . . . . . . . . . .  12
     10.1.  Normative References . . . . . . . . . . . . . . . . . .  12
     10.2.  Informative References . . . . . . . . . . . . . . . . .  12














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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.  This
   loss might be due to congestion; it might also be a result of an
   unplanned outage caused by a flapping link, a 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 is left up to the RTP
   layer and the applications to detect, and recover from, packet loss.

   In a carefully managed network, congestion should not normally
   happen; however, network outages can still happen due to the reasons
   listed above.  In such a managed network, one technique to recover
   from packet loss without incurring unbounded delay is to duplicate
   the packets and send them in separate redundant streams.  As
   described later in this document, the probability that two copies of
   the same packet are lost in cases of non-congestive packet loss is
   quite small.

   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 that 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 overhead in terms of
   bandwidth, since everything is sent twice, but with a low overhead in
   terms of processing.  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.








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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.  Use Cases for Dual Streaming

   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 (in addition to the
   transport ports), since they will be both routed over the same path:

   o  IP Source Address

   o  IP Destination Address

   Two routing-plane identical RTP streams might carry the same payload
   but can use different Synchronization Sources (SSRCs) 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 [RFC7197].  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





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   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, the destination host could also have multiple IP
   addresses for an RTP source to send the redundant streams to.

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 flow
       labels will be the same.  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 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
       are often small.  This is what is also known as "time-shifted
       redundancy", "temporal redundancy" or simply "delayed
       duplication" [RFC7197] [IC2011].  This approach can be used with
       both types of dual streaming, described in Sections 3.1 and 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



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       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
       ensured 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).

   Temporal and spatial redundancy deal with different patterns of
   packet loss.  The former helps with transient loss (within the
   duplication window), while the latter helps with longer-term packet
   loss that affects only one of the two redundant paths.

   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 currently 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.

   When the original and duplicate RTP streams are carried in separate
   "m" lines in a Session Description Protocol (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; this is left for future standardization.








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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 and flow label might also
   cause spatial redundancy (which might introduce an additional delay
   due to the delta between the path delays), and so it 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 Section 8 of
   [RFC3550].

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 mechanism based on RTCP Sender Reports, 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

   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



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   which stream is the duplicate.  (The latter can be done in-band using
   an RTCP extension or out-of-band in the SDP description.)

   Out-of-band signaling is needed for both features.  The SDP attribute
   to signal duplication in the SDP offer/answer exchange ('duplication-
   delay') is defined in [RFC7197].  The required SDP grouping semantics
   are defined in [RFC7104].

   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

   Section 3.2 of [RFC7104] states that 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.

5.  Use of RTP and RTCP with Spatial Redundancy

   Assuming the network is structured appropriately, 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 from 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.




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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 mechanism based on RTCP Sender Reports, 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 [RFC7104].
   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
        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





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6.  Use of RTP and RTCP with Temporal and Spatial Redundancy

   This uses the same RTP/RTCP mechanisms from Sections 4 and 5, plus a
   combination of signaling provided in each of these sections.

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 (e.g., through
   explicit operator configuration) 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 take
   place inside the network before the ultimate endpoints that will
   consume the RTP media, the whole process is transparent to these
   endpoints.  Thus, these endpoints will apply any congestion control,
   if applicable, on the de-duplicated RTP stream.  This output stream
   will have fewer losses than either the original or duplicated stream
   will have, and the endpoint will make congestion control decisions
   accordingly.  However, if de-duplication takes place at the ultimate
   endpoint, this endpoint MUST consider the aggregate of the original
   and duplicated RTP stream in any congestion control it wants to
   apply.  The endpoint 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.  In these scenarios, the RTP Monitoring Framework [RFC6792]
   can be used to monitor the duplicated streams in the same way an
   ordinary RTP would be monitored.





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8.  Security Considerations

   The security considerations of [RFC3550], [RFC7104], [RFC7197], 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 Secure RTP (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 such middleboxes 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 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 setup signaling.  Details of how that is done will
   depend on the type of signaling used (SIP, Real Time Streaming
   Protocol (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 to
   have access to the appropriate signaling and keying material.

   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.  Acknowledgments

   Thanks to Magnus Westerlund for his suggestions.









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10.  References

10.1.  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.

   [RFC7197]  Begen, A., Cai, Y., and H. Ou, "Duplication Delay
              Attribute in the Session Description Protocol", RFC 7197,
              April 2014.

   [RFC7104]  Begen, A., Cai, Y., and H. Ou, "Duplication Grouping
              Semantics in the Session Description Protocol", RFC 7104,
              January 2014.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, March 2004.

10.2.  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.

   [RFC6363]  Watson, M., Begen, A., and V. Roca, "Forward Error
              Correction (FEC) Framework", RFC 6363, October 2011.

   [RFC6792]  Wu, Q., Hunt, G., and P. Arden, "Guidelines for Use of the
              RTP Monitoring Framework", RFC 6792, November 2012.

   [IC2011]   Evans, J., Begen, A., Greengrass, J., and C. Filsfils,
              "Toward Lossless Video Transport", IEEE Internet
              Computing, Vol. 15, No. 6, pp. 48-57, November 2011.










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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://csperkins.org/
































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