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rfc:rfc7198

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.

Begen & Perkins Standards Track [Page 1] RFC 7198 RTP Duplication April 2014

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

Begen & Perkins Standards Track [Page 2] RFC 7198 RTP Duplication April 2014

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.

Begen & Perkins Standards Track [Page 3] RFC 7198 RTP Duplication April 2014

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

Begen & Perkins Standards Track [Page 4] RFC 7198 RTP Duplication April 2014

 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

Begen & Perkins Standards Track [Page 5] RFC 7198 RTP Duplication April 2014

     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.

Begen & Perkins Standards Track [Page 6] RFC 7198 RTP Duplication April 2014

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

Begen & Perkins Standards Track [Page 7] RFC 7198 RTP Duplication April 2014

 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.

Begen & Perkins Standards Track [Page 8] RFC 7198 RTP Duplication April 2014

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

Begen & Perkins Standards Track [Page 9] RFC 7198 RTP Duplication April 2014

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.

Begen & Perkins Standards Track [Page 10] RFC 7198 RTP Duplication April 2014

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.

Begen & Perkins Standards Track [Page 11] RFC 7198 RTP Duplication April 2014

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.

Begen & Perkins Standards Track [Page 12] RFC 7198 RTP Duplication April 2014

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/

Begen & Perkins Standards Track [Page 13]

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