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Versions: (draft-xia-avtext-splicing-for-rtp) 00 01 02 03 04 05 06 07 08 09 10 11 12 13 RFC 6828

AVTEXT Working Group                                              J. Xia
Internet-Draft                                                    Huawei
Intended status: Informational                         November 14, 2012
Expires: May 18, 2013


                   Content Splicing for RTP Sessions
                 draft-ietf-avtext-splicing-for-rtp-13

Abstract

   Content splicing is a process that replaces the content of a main
   multimedia stream with other multimedia content, and delivers the
   substitutive multimedia content to the receivers for a period of
   time.  Splicing is commonly used for local advertisement insertion by
   cable operators, replacing a national advertisement content with a
   local advertisement.

   This memo describes some use cases for content splicing and a set of
   requirements for splicing content delivered by RTP.  It provides
   concrete guidelines for how an RTP mixer can be used to handle
   content splicing.

Status of this Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on May 18, 2013.

Copyright Notice

   Copyright (c) 2012 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



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   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.  System Model and Terminology . . . . . . . . . . . . . . . . .  3
   3.  Requirements for RTP Splicing  . . . . . . . . . . . . . . . .  6
   4.  Content Splicing for RTP sessions  . . . . . . . . . . . . . .  7
     4.1.  RTP Processing in RTP Mixer  . . . . . . . . . . . . . . .  7
     4.2.  RTCP Processing in RTP Mixer . . . . . . . . . . . . . . .  8
     4.3.  Considerations for Handling Media Clipping at the RTP
           Layer  . . . . . . . . . . . . . . . . . . . . . . . . . . 10
     4.4.  Congestion Control Considerations  . . . . . . . . . . . . 11
     4.5.  Considerations for Implementing Undetectable Splicing  . . 12
   5.  Implementation Considerations  . . . . . . . . . . . . . . . . 13
   6.  Security Considerations  . . . . . . . . . . . . . . . . . . . 13
   7.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 14
   8.  Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 14
   9.  10. Appendix- Why Mixer Is Chosen  . . . . . . . . . . . . . . 15
   10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 15
     10.1. Normative References . . . . . . . . . . . . . . . . . . . 15
     10.2. Informative References . . . . . . . . . . . . . . . . . . 16
   Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 16






















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1.  Introduction

   This document outlines how content splicing can be used in RTP
   sessions.  Splicing, in general, is a process where part of a
   multimedia content is replaced with other multimedia content, and
   delivered to the receivers for a period of time.  The substitutive
   content can be provided for example via another stream or via local
   media file storage.  One representative use case for splicing is
   local advertisement insertion, allowing content providers to replace
   the national advertising content with its own regional advertising
   content prior to delivering the regional advertising content to the
   receivers.  Besides the advertisement insertion use case, there are
   other use cases in which splicing technology can be applied.  For
   example, splicing a recorded video into a video conferencing session,
   or implementing a playlist server that stitches pieces of video
   together.

   Content splicing is a well-defined operation in MPEG-based cable TV
   systems.  Indeed, the Society for Cable Telecommunications Engineers
   (SCTE) has created two standards, [SCTE30] and [SCTE35], to
   standardize MPEG2-TS splicing procedure.  SCTE 30 creates a
   standardized method for communication between advertisements server
   and splicer, and SCTE 35 supports splicing of MPEG2 transport
   streams.

   When using multimedia splicing into the internet, the media may be
   transported by RTP.  In this case the original media content and
   substitutive media content will use the same time period, but may
   contain different numbers of RTP packets due to different media
   codecs and entropy coding.  This mismatch may require some
   adjustments of the RTP header sequence number to maintain
   consistency.  [RFC3550] provides the tools to enabled seamless
   content splicing in RTP session, but to date there has been no clear
   guidelines on how to use these tools.

   This memo outlines the requirements for content splicing in RTP
   sessions and describes how an RTP mixer can be used to meet these
   requirements.


2.  System Model and Terminology

   In this document, an intermediary network element, the Splicer
   handles RTP splicing.  The Splicer can receive main content and
   substitutive content simultaneously, but will send one of them at one
   point of time.

   When RTP splicing begins, the splicer sends the substitutive content



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   to the RTP receiver instead of the main content for a period of time.
   When RTP splicing ends, the splicer switches back sending the main
   content to the RTP receiver.

   A simplified RTP splicing diagram is depicted in Figure 1, in which
   only one main content flow and one substitutive content flow are
   given.  Actually, the splicer can handle multiple splicing for
   multiple RTP sessions simultaneously.  RTP splicing may happen more
   than once in multiple time slots during the lifetime of the main RTP
   stream.  The methods how splicer learns when to start and end the
   splicing is out of scope for this document.


      +---------------+
      |               | Main Content +-----------+
      |   Main RTP    |------------->|           | Output Content
      |   Content     |              |  Splicer  |--------------->
      +---------------+   ---------->|           |
                         |           +-----------+
                         |
                         | Substitutive Content
                         |
                         |
               +-----------------------+
               |   Substitutive RTP    |
               |       Content         |
               |          or           |
               |   Local File Storage  |
               +-----------------------+


               Figure 1: RTP Splicing Architecture


   This document uses the following terminologies.

   Output RTP Stream

      The RTP stream that the RTP receiver is currently receiving.  The
      content of output RTP stream can be either main content or
      substitutive content.

   Main Content

      The multimedia content that are conveyed in main RTP stream.  Main
      content will be replaced by the substitutive content during
      splicing.




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   Main RTP Stream

      The RTP stream that the splicer is receiving.  The content of main
      RTP stream can be replaced by substitutive content for a period of
      time.

   Main RTP Sender

      The sender of RTP packets carrying the main RTP stream.

   Substitutive Content

      The multimedia content that replaces the main content during
      splicing.  The substitutive content can for example be contained
      in an RTP stream from a media sender or fetched from local media
      file storage.

   Substitutive RTP Stream

      A RTP stream with new content that will replace the content in the
      main RTP stream.  Substitutive RTP stream and main RTP stream are
      two separate streams.  If the substitutive content is provided via
      substitutive RTP stream, the substitutive RTP Stream must pass
      through the splicer before the substitutive content is delivered
      to receiver.

   Substitutive RTP Sender

      The sender of RTP packets carrying the substitutive RTP stream.

   Splicing In Point

      A virtual point in the RTP stream, suitable for substitutive
      content entry, typically in the boundary between two independently
      decodable frames.

   Splicing Out Point

      A virtual point in the RTP stream, suitable for substitutive
      content exist, typically in the boundary between two independently
      decodable frames.

   Splicer

      An intermediary node that inserts substitutive content into main
      RTP stream.  The splicer sends substitutive content to RTP
      receiver instead of main content during splicing.  It is also
      responsible for processing RTCP traffic between the RTP sender and



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      the RTP receiver.


3.  Requirements for RTP Splicing

   In order to allow seamless content splicing at the RTP layer, the
   following requirements must be met.  Meeting these will also allow,
   but not require, seamless content splicing at layers above RTP.

   REQ-1:

      The splicer should be agnostic about the network and transport
      layer protocols used to deliver the RTP streams.

   REQ-2:

      The splicing operation at the RTP layer must allow splicing at any
      point required by the media content, and must not constrain when
      splicing in or splicing out operations can take place.

   REQ-3:

      Splicing of RTP content must be backward compatible with the RTP/
      RTCP protocol, associated profiles, payload formats, and
      extensions.

   REQ-4:

      The splicer will modify the content of RTP packets, and thus break
      the end-to-end security, at a minimum breaking the data integrity
      and source authentication.  If the Splicer is designated to insert
      substitutive content, it must be trusted, i.e., be in the security
      context(s) with the main RTP sender, the substitutive RTP sender,
      and the receivers.  If encryption is employed, the splicer
      commonly must decrypt the inbound RTP packets and re-encrypt the
      outbound RTP packets after splicing.

   REQ-5:

      The splicer should rewrite as necessary and forward RTCP messages
      (e.g., including packet loss, jitter, etc.) sent from downstream
      receiver to the main RTP sender or the substitutive RTP sender,
      and thus allow the main RTP sender or substitutive RTP sender to
      learn the performance of the downstream receiver when its content
      is being passed to RTP receiver.  In addition, the splicer should
      rewrite RTCP messages from the main RTP sender or substitutive RTP
      sender to the receiver.




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   REQ-6:

      The splicer must not affect other RTP sessions running between the
      RTP sender and the RTP receiver, and must be transparent for the
      RTP sessions it does not splice.

   REQ-7:

      The RTP receiver should not be able to detect any splicing points
      in the RTP stream produced by the splicer on RTP protocol level.
      For the advertisement insertion use case, it is important to make
      it difficult for the RTP receiver to detect where an advertisement
      insertion is starting or ending from the RTP packets, and thus
      avoiding the RTP receiver from filtering out the advertisement
      content.  This memo only focuses on making the splicing
      undetectable at the RTP layer.  The corresponding processing is
      depicted in section 4.5.



4.  Content Splicing for RTP sessions

   The RTP specification [RFC3550] defines two types of middlebox: RTP
   translators and RTP mixers.  Splicing is best viewed as a mixing
   operation.  The splicer generates a new RTP stream that is a mix of
   the main RTP stream and the substitutive RTP stream.  An RTP mixer is
   therefore an appropriate model for a content splicer.  In next four
   subsections (from subsection 4.1 to subsection 4.4), the document
   analyzes how the mixer handles RTP splicing and how it satisfies the
   general requirements listed in section 3.  In subsection 4.5, the
   document looks at REQ-7 in order to hide the fact that splicing take
   place.

4.1.  RTP Processing in RTP Mixer

   A splicer could be implemented as a mixer that receives the main RTP
   stream and the substitutive content (possibly via a substitutive RTP
   stream), and sends a single output RTP stream to the receiver(s).
   That output RTP stream will contain either the main content or the
   substitutive content.  The output RTP stream will come from the
   mixer, and will have the synchronization source (SSRC) of the mixer
   rather than the main RTP sender or the substitutive RTP sender.

   The mixer uses its own SSRC, sequence number space and timing model
   when generating the output stream.  Moreover, the mixer may insert
   the SSRC of main RTP stream into contributing source (CSRC) list in
   the output media stream.




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   At the splicing in point, when the substitutive content becomes
   active, the mixer chooses the substitutive RTP stream as input stream
   at splicing in point, and extracts the payload data (i.e.,
   substitutive content).  If the substitutive content comes from local
   media file storage, the mixer directly fetches the substitutive
   content.  After that, the mixer encapsulates substitutive content
   instead of main content as the payload of the output media stream,
   and then sends the output RTP media stream to receiver.  The mixer
   may insert the SSRC of substitutive RTP stream into CSRC list in the
   output media stream.  If the substitutive content comes from local
   media file storage, the mixer should leave the CSRC list blank.

   At the splicing out point, when the substitutive content ends, the
   mixer retrieves the main RTP stream as input stream at splicing out
   point, and extracts the payload data (i.e., main content).  After
   that, the mixer encapsulates main content instead of substitutive
   content as the payload of the output media stream, and then sends the
   output media stream to the receivers.  Moreover, the mixer may insert
   the SSRC of main RTP stream into CSRC list in the output media stream
   as before.

   Note that if the content is too large to fit into RTP packets sent to
   RTP receiver, the mixer needs to transcode or perform application-
   layer fragmentation.  Usually the mixer is deployed as part of a
   managed system and MTU will be carefully managed by this system.
   This document does not raise any new MTU related issues compared to a
   standard mixer described in [RFC3550].

   Splicing may occur more than once during the lifetime of main RTP
   stream, this means the mixer needs to send main content and
   substitutive content in turn with its own SSRC identifier.  From
   receiver point of view, the only source of the output stream is the
   mixer regardless of where the content is coming from.

4.2.  RTCP Processing in RTP Mixer

   By monitoring available bandwidth and buffer levels and by computing
   network metrics such as packet loss, network jitter, and delay, RTP
   receiver can learn the network performance and communicate this to
   the RTP sender via RTCP reception reports.

   According to the description in section 7.3 of [RFC3550], the mixer
   splits the RTCP flow between sender and receiver into two separate
   RTCP loops, RTP sender has no idea about the situation on the
   receiver.  But splicing is a processing that the mixer selects one
   media stream from multiple streams rather than mixing them, so the
   mixer can leave the SSRC identifier in the RTCP report intact (i.e.,
   the SSRC of downstream receiver), this enables the main RTP sender or



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   the substitutive RTP sender to learn the situation on the receiver.

   If the RTCP report corresponds to a time interval that is entirely
   main content or entirely substitutive content, the number of output
   RTP packets containing substitutive content is equal to the number of
   input substitutive RTP packets (from substitutive RTP stream) during
   splicing, in the same manner, the number of output RTP packets
   containing main content is equal to the number of input main RTP
   packets (from main RTP stream) during non-splicing unless the mixer
   fragment the input RTP packets.  This means that the mixer does not
   need to modify the loss packet fields in reception report blocks in
   RTCP reports.  But if the mixer fragments the input RTP packets, it
   may need to modify the loss packet fields to compensate for the
   fragmentation.  Whether the input RTP packets are fragmented or not,
   the mixer still needs to change the SSRC field in report block to the
   SSRC identifier of the main RTP sender or the substitutive RTP
   sender, and rewrite the extended highest sequence number field to the
   corresponding original extended highest sequence number before
   forwarding the RTCP report to the main RTP sender or the substitutive
   RTP sender.

   If the RTCP report spans the splicing in point or the splicing out
   point, it reflects the characteristics of the combination of main RTP
   packets and substitutive RTP packets.  In this case, the mixer needs
   to divide the RTCP report into two separate RTCP reports and send
   them to their original RTP senders respectively.  For each RTCP
   report, the mixer also needs to make the corresponding changes to the
   packet loss fields in report block besides the SSRC field and the
   extended highest sequence number field.

   If the mixer receives an RTCP extended report (XR) block, it should
   rewrite the XR report block in a similar way to the reception report
   block in the RTCP report.

   Besides forwarding the RTCP reports sent from RTP receiver, the mixer
   can also generate its own RTCP reports to inform the main RTP sender
   or the substitutive RTP sender of the reception quality of the
   content reaches the mixer when the content is not sent to the RTP
   receiver.  These RTCP reports use the SSRC of the mixer.  If the
   substitutive content comes from local media file storage, the mixer
   does not need to generate RTCP reports for the substitutive stream.

   Based on above RTCP operating mechanism, the RTP sender whose content
   is being passed to receiver will see the reception quality of its
   stream as received by the mixer, and the reception quality of spliced
   stream as received by the receiver.  The RTP sender whose content is
   not being passed to receiver will only see the reception quality of
   its stream as received by the mixer.



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   The mixer must forward RTCP SDES and BYE packets from the receiver to
   the sender, and may forward them in inverse direction as defined in
   section 7.3 of [RFC3550].

   Once the mixer receives an RTP/AVPF [RFC4585] transport layer
   feedback packet, it must handle it carefully as the feedback packet
   may contain the information of the content that come from different
   RTP senders.  In this case the mixer needs to divide the feedback
   packet into two separate feedback packets and process the information
   in the feedback control information (FCI) in the two feedback
   packets, just as the RTCP report process described above.

   If the substitutive content comes from local media file storage
   (i.e., the mixer can be regarded as the substitutive RTP sender), any
   RTCP packets received from downstream relate to the substitutive
   content must be terminated on the mixer without any further
   processing.

4.3.  Considerations for Handling Media Clipping at the RTP Layer

   This section provides informative guidelines on how to handle media
   substitution at both the RTP layer to minimize media impact.  Dealing
   with the media substitution well at the RTP layer is necessary for
   quality implementations.  To perfectly erase any media impact needs
   more considerations at the higher layers, how the media substitution
   is erased at the higher layers are outside of the scope of this memo.

   If the time duration for any substitutive content mismatches, i.e.,
   shorter or longer, than the duration of the main content to be
   replaced, then media degradations may occur at the splicing point and
   thus impact the user's experience.

   If the substitutive content has shorter duration from the main
   content, then there could be a gap in the output RTP stream.  The RTP
   sequence number will be contiguous across this gap, but there will be
   an unexpected jump in the RTP timestamp.  Such a gap would cause the
   receiver to have nothing to play.  This may be unavoidable, unless
   the mixer can adjusts the splice in or splice out point to
   compensate.  This assumes the splicing mixer can send more of the
   main RTP stream in place of the shorter substitutive stream, or vary
   the length of the substitutive content.  It is the responsibility of
   the higher layer protocols and the media providers to ensure that the
   substitutive content is of very similar duration as the main content
   to be replaced.

   If the substitute content has longer duration than the reserved gap
   duration, there will be an overlap between the substitutive RTP
   stream and the main RTP stream at the splicing out point.  A



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   straightforward approach is that the mixer performs an ungraceful
   action, terminating the splicing and switching back to main RTP
   stream even if this may cause media stuttering on receiver.
   Alternatively, the mixer may transcode the substitutive content to
   play at a faster rate than normal, to adjust it to the length of the
   gap in the main content, and generate a new RTP stream for the
   transcoded content.  This is a complex operation, and very specific
   to the content and media codec used.  Additional approaches exists,
   these types of issues should be taken into account in both mixer
   implementors and media generators to enable smooth substitutions.

4.4.  Congestion Control Considerations

   If the substitutive content has somewhat different characteristics
   from the main content it replaces, or if the substitutive content is
   encoded with a different codec or has different encoding bitrate, it
   might overload the network and might cause network congestion on the
   path between the mixer and the RTP receiver(s) that would not have
   been caused by the main content.

   To be robust to network congestion and packet loss, a mixer that is
   performing splicing must continuously monitor the status of
   downstream network by monitoring any of the following RTCP reports
   that are used:

   1.  RTCP receiver reports indicate packet loss [RFC3550].

   2.  RTCP NACKs for lost packet recovery [RFC4585].

   3.  RTCP ECN Feedback information [RFC6679].

   Once the mixer detects congestion on its downstream link, it will
   treat these reports as follows:

   1.  If the mixer receives the RTCP receiver reports with packet loss
       indication, it will forward the reports to the substitutive RTP
       sender or the main RTP sender as described in section 4.2.

   2.  If mixer receives the RTCP NACK packets defined in [RFC4585] from
       RTP receiver for packet loss recovery, it first identifies the
       content category of lost packets to which the NACK corresponds.
       Then, the mixer will generate new RTCP NACK for the lost packets
       with its own SSRC, and make corresponding changes to their
       sequence numbers to match original, pre-spliced, packets.  If the
       lost substitutive content comes from local media file storage,
       the mixer acting as substitutive RTP sender will directly fetch
       the lost substitutive content and retransmit it to RTP receiver.
       The mixer may buffer the sent RTP packets and do the



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       retransmission.

       It is somewhat complex that the lost packets requested in a
       single RTCP NACK message not only contain the main content but
       also the substitutive content.  To address this, the mixer must
       divide the RTCP NACK packet into two separate RTCP NACK packets:
       one requests for the lost main content, and another requests for
       the lost substitutive content.

   3.  If an ECN-aware mixer receives RTCP ECN feedbacks (RTCP ECN
       feedback packets or RTCP XR summary reports) defined in [RFC6679]
       from the RTP receiver, it must process them in a similar way to
       the RTP/AVPF feedback packet or RTCP XR process described in
       section 4.2 of this memo.

   These three methods require the mixer to run a congestion control
   loop and bitrate adaptation between itself and RTP receiver.  The
   mixer can thin or transcode the main RTP stream or the substitutive
   RTP stream, but such operations are very inefficient and difficult,
   and bring undesirable delay.  Fortunately in this memo, the mixer
   acting as splicer can rewrite the RTCP packets sent from the RTP
   receiver and forward them to the RTP sender, thus letting the RTP
   sender knows that congestion is being experienced on the path between
   the mixer and the RTP receiver.  Then, the RTP sender applies its
   congestion control algorithm and reduces the media bitrate to a value
   that is in compliance with congestion control principles for the
   slowest link.  The congestion control algorithm may be a TCP-friendly
   bitrate adaptation algorithm specified in [RFC5348], or a DCCP
   congestion control algorithms defined in [RFC5762].

   If the substitutive content comes from local media file storage, the
   mixer must directly reduce the bitrate as if it were the substitutive
   RTP sender.

   From above analysis, to reduce the risk of congestion and remain the
   bandwidth consumption stable over time, the substitutive RTP stream
   is recommended to be encoded at an appropriate bitrate to match that
   of main RTP stream.  If the substitutive RTP stream comes from the
   substitutive RTP sender, this sender had better has some knowledge
   about the media encoding bitrate of main content in advance.  How it
   knows that is out of scope in this draft.

4.5.  Considerations for Implementing Undetectable Splicing

   If it is desirable to prevent receivers from detecting that splicing
   is occurring at the RTP layer, the mixer must not include a CSRC list
   in outgoing RTP packets, and must not forward RTCP messages from the
   main RTP sender or from the substitutive RTP sender.  Due to the



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   absence of CSRC list in the output RTP stream, the RTP receiver only
   initiates SDES, BYE and APP packets to the mixer without any
   knowledge of the main RTP sender and the substitutive RTP sender.

   CSRC list identifies the contributing sources, these SSRC identifiers
   of contributing sources are kept globally unique for each RTP
   session.  The uniqueness of SSRC identifier is used to resolve
   collisions and detecting RTP-level forwarding loops as defined in
   section 8.2 of [RFC3550].  The absence of CSRC list in this case will
   create a danger that loops involving those contributing sources could
   not be detected.  The loops could occur if either the mixer is
   misconfigured to form a loop, or a second mixer/translator is added,
   causing packets to loop back to upstream of the original mixer.  An
   undetected RTP packet loop is a serious denial of service threat,
   which can consume all available bandwidth or mixer processing
   resources until the looped packets are dropped as result of
   congestion.  So Non-RTP means must be used to detect and resolve
   loops if the mixer does not add a CSRC list.


5.  Implementation Considerations

   When the mixer is used to handle RTP splicing, RTP receiver does not
   need any RTP/RTCP extension for splicing.  As a trade-off, additional
   overhead could be induced on the mixer which uses its own sequence
   number space and timing model.  So the mixer will rewrite RTP
   sequence number and timestamp whatever splicing is active or not, and
   generate RTCP flows for both sides.  In case the mixer serves
   multiple main RTP streams simultaneously, this may lead to more
   overhead on the mixer.

   If undetectable splicing requirement is required, CSRC list is not
   included in outgoing RTP packet, this brings a potential issue with
   loop detection as briefly described in section 4.5.


6.  Security Considerations

   The splicing application is subject to the general security
   considerations of the RTP specification [RFC3550].

   The mixer acting as splicer replaces some content with other content
   in RTP packets, thus breaking any RTP level end-to-end security, such
   as integrity protection and source authentication.  Thus any RTP
   level or outside security mechanism, such as IPSec or DTLS will use a
   security association between the splicer and the receiver.  When
   using SRTP the splicer could be provisioned with the same security
   association as the main RTP sender.  Using a limitation in the SRTP



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   security services regarding source authentication, the splicer can
   modify and re-protect the RTP packets without enabling the receiver
   to detect if the data comes from the original source or from the
   splicer.

   Security goals to have source authentication all the way from the RTP
   main sender to the receiver through the splicer is not possible with
   splicing and any existing solutions.  A new solution can
   theoretically be developed that enables identifying the participating
   entities and what each provides, i.e. the different media sources,
   main and substituting, and the splicer providing the RTP level
   integration of the media payloads in a common timeline and
   synchronization context.  Such a solution would obviously not meet
   Req-7 and will be detectable on RTP level.

   The nature of this RTP service offered by a network operator
   employing a content splicer is that the RTP layer security
   relationship is between the receiver and the splicer, and between the
   senders and the splicer, are not end-to-end.  This appears to
   invalidate the undetectability goal, but in the common case the
   receiver will consider the splicer as the main media source.

   Some RTP deployments use RTP payload security mechanisms (e.g.,
   ISMACryp [ISMACryp]).  If any payload internal security mechanisms
   are used, only the RTP sender and the RTP receiver establish that
   security context, in which case, any middlebox (e.g., splicer)
   between the RTP sender and the RTP receiver will not get such keying
   material.  This may impact the splicer's possibility to perform
   splicing if it is dependent on RTP payload level hints for finding
   the splice in and out points.  However, other potential solutions
   exist to specify or mark where the splicing points exist in the media
   streams.  When using RTP payload security mechanisms SRTP or other
   security mechanism at RTP or lower layers can be used to provide
   integrity and source authentication between the splicer and the RTP
   receiver.


7.  IANA Considerations

   No IANA actions are required.


8.  Acknowledgments

   The following individuals have reviewed the earlier versions of this
   specification and provided very valuable comments: Colin Perkins,
   Magnus Westerlund, Roni Even, Tom Van Caenegem, Joerg Ott, David R
   Oran, Cullen Jennings, Ali C Begen, Charles Eckel and Ning Zong.



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9.  10. Appendix- Why Mixer Is Chosen

   Translator and mixer both can realize splicing by changing a set of
   RTP parameters.

   Translator has no SSRC, hence it is transparent to RTP sender and
   receiver.  Therefore, RTP sender sees the full path to the receiver
   when translator is passing its content.  When translator insert the
   substitutive content RTP sender could get a report on the path up to
   translator itself.  Additionally, if splicing does not occur yet,
   translator does not need to rewrite RTP header, the overhead on
   translator can be avoided.

   If mixer is used to do splicing, it can also allow RTP sender to
   learn the situation of its content on receiver or on mixer just like
   translator does, which is specified in section 4.2.  Compared to
   translator, mixer's outstanding benefit is that it is pretty straight
   forward to do with RTCP messages, for example, bit-rate adaptation to
   handle varying network conditions.  But translator needs more
   considerations and its implementation is more complex.

   From above analysis, both translator and mixer have their own
   advantages: less overhead or less complexity on handling RTCP.
   Through long and sophisticated discussion, the avtext WG members
   prefer less complexity rather than less overhead and incline to mixer
   to do splicing.

   If one chooses mixer as splicer, the overhead on mixer must be taken
   into account even if the splicing does not occur yet.


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.

   [RFC4585]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
              "Extended RTP Profile for Real-time Transport Control
              Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
              July 2006.

   [RFC6679]  Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P.,
              and K. Carlberg, "Explicit Congestion Notification (ECN)
              for RTP over UDP", RFC 6679, August 2012.




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10.2.  Informative References

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

   [RFC5348]  Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
              Friendly Rate Control (TFRC): Protocol Specification",
              RFC 5348, September 2008.

   [RFC5762]  Perkins, C., "RTP and the Datagram Congestion Control
              Protocol (DCCP)", RFC 5762, April 2010.

   [SCTE30]   Society of Cable Telecommunications Engineers (SCTE),
              "Digital Program Insertion Splicing API", 2009.

   [SCTE35]   Society of Cable Telecommunications Engineers (SCTE),
              "Digital Program Insertion Cueing Message for Cable",
              2011.

   [ISMACryp]
              Internet Streaming Media Alliance (ISMA), "ISMA Encryption
              and Authentication Specification 2.0", November 2007.


Author's Address

   Jinwei Xia
   Huawei
   Software No.101
   Nanjing, Yuhuatai District  210012
   China

   Phone: +86-025-86622310
   Email: xiajinwei@huawei.com
















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