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Versions: (draft-xia-avtext-splicing-notification) 00 01 02 03 04 05 06 07 08 09 RFC 8286

AVTEXT Working Group                                              J. Xia
INTERNET-DRAFT                                                   R. Even
Intended Status: Standards Track                                R. Huang
Expires: December 30, 2016                                        Huawei
                                                                 L. Deng
                                                            China Mobile
                                                           June 28, 2016


            RTP/RTCP extension for RTP Splicing Notification
               draft-ietf-avtext-splicing-notification-08


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. The splicer is designed to handle RTP splicing and needs to
   know when to start and end the splicing.

   This memo defines two RTP/RTCP extensions to indicate the splicing
   related information to the splicer: an RTP header extension that
   conveys the information in-band and an RTCP packet that conveys the
   information out-of-band.

Status of this Memo

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

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
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   http://www.ietf.org/shadow.html





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Copyright and License Notice

   Copyright (c) 2016 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
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   include Simplified BSD License text as described in Section 4.e of
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   described in the Simplified BSD License.



Table of Contents

   1  Introduction  . . . . . . . . . . . . . . . . . . . . . . . . .  3
     1.1  Terminology . . . . . . . . . . . . . . . . . . . . . . . .  3
   2  Overview  . . . . . . . . . . . . . . . . . . . . . . . . . . .  4
     2.1 Overview of RTP Splicing . . . . . . . . . . . . . . . . . .  4
     2.2 Overview of Splicing Interval  . . . . . . . . . . . . . . .  5
   3  Conveying Splicing Interval in RTP/RTCP extensions  . . . . . .  5
     3.1 RTP Header Extension . . . . . . . . . . . . . . . . . . . .  5
     3.2 RTCP Splicing Notification Message . . . . . . . . . . . . .  6
   4  Reducing Splicing Latency . . . . . . . . . . . . . . . . . . .  7
   5  Failure Cases . . . . . . . . . . . . . . . . . . . . . . . . .  8
   6 SDP Signaling  . . . . . . . . . . . . . . . . . . . . . . . . .  8
     6.1 Declarative SDP  . . . . . . . . . . . . . . . . . . . . . .  9
     6.2 Offer/Answer without BUNDLE  . . . . . . . . . . . . . . . .  9
     6.3 Offer/Answer with BUNDLE: All Media are spliced  . . . . . . 10
     6.4 Offer/Answer with BUNDLE: a Subset of Media are Spliced  . . 12
   7  Security Considerations . . . . . . . . . . . . . . . . . . . . 13
   8  IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 14
     8.1  RTCP Control Packet Types . . . . . . . . . . . . . . . . . 14
     8.2  RTP Compact Header Extensions . . . . . . . . . . . . . . . 14
     8.3 SDP Grouping Semantic Extension  . . . . . . . . . . . . . . 14
   9 Acknowledges . . . . . . . . . . . . . . . . . . . . . . . . . . 15
   10  References . . . . . . . . . . . . . . . . . . . . . . . . . . 15
     10.1  Normative References . . . . . . . . . . . . . . . . . . . 15
     10.2  Informative References . . . . . . . . . . . . . . . . . . 15
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 16







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1  Introduction

   Splicing is a process that replaces some multimedia content with
   other multimedia content and delivers the substitutive multimedia
   content to the receivers for a period of time. In some predictable
   splicing cases, e.g., advertisement insertion, the splicing duration
   needs to be inside of the specific, pre-designated time slot. Certain
   timing information about when to start and end the splicing must be
   first acquired by the splicer in order to start the splicing. This
   document refers to this information as the Splicing Interval.

   [SCTE35] provides a method that encapsulates the Splicing Interval
   inside the MPEG2-TS layer in cable TV systems. When transported in
   RTP, an middle box designed as the splicer to decode the RTP packets
   and search for the Splicing Interval inside the payloads is required.
   The need for such processing increases the workload of the middle box
   and limits the number of RTP sessions the middle box can support.

   The document defines an RTP header extension [RFC5285] used by the
   main RTP sender to provide the Splicing Interval by including it in
   the RTP packets.

   However, the Splicing Interval conveyed in the RTP header extension
   might not reach the splicer successfully. Any splicing un-aware
   middlebox on the path between the RTP sender might strip this RTP
   header extension.

   To increase robustness against such case, the document also defines a
   new RTCP packet type to carry the same Splicing Interval to the
   splicer. Since RTCP is also unreliable and may not be so immediate as
   the in-band way, it's only considered as a complement to the RTP
   header extension.

1.1  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC 2119 [RFC2119].

   In addition, we define following terminologies:

   Main RTP sender:

      The sender of RTP packets carrying the main RTP stream.

   Splicer:

      An intermediary node that inserts substitutive content into a main



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

   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 exit, typically in the boundary between two independently
      decodable frames.

   Splicing Interval:

      The NTP-format timestamps, representing the main RTP sender
      wallclock time, for the Splicing-In point and Splicing-Out point
      per [RFC6828] allowing the splicer to know when to start and end
      the RTP splicing.

   Substitutive RTP Sender:

      The sender of RTP packets carrying the RTP stream that will
      replace the content in the main RTP stream.

2  Overview

2.1 Overview of RTP Splicing

   RTP Splicing is intended to replace some multimedia content with
   certain substitutive multimedia content, and then forward it to the
   receivers for a period of time. This process is authorized by the
   main RTP sender that offers a specific time window for inserting the
   substitutive multimedia content in the main content. A typical usage
   is that IPTV service provider uses its own regional advertising
   content to replace national advertising content, the time window of
   which is explicitly indicated by the IPTV service provider.

   The splicer is a middlebox handling RTP splicing. It receives main
   content and substitutive content simultaneously but only chooses to
   send one of them to the receiver at any point of time. When RTP
   splicing begins, the splicer sends the substitutive content to the
   receivers instead of the main content. When RTP splicing ends, the
   splicer switches back to sending the main content to the receivers.



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   This implies that the receiver is explicitly configured to receive
   the traffic via the splicer, and will return any RTCP feedback to it
   in the presence of the splicer.

   The middlebox working as the splicer can be implemented as either an
   RTP mixer or as an RTP translator. If implemented as an RTP mixer,
   [RFC6828] specifies how the splicer can use its own SSRC, sequence
   number space, and timing model when generating the output stream to
   receivers, using the CSRC list to indicate whether the original or
   substitutive content is being delivered. The splicer, on behalf of
   the content provider, can omit the CSRC list from the RTP packets it
   generates. This simplifies the design of the receivers, since they
   don't need to parse the CSRC list, but makes it harder to determine
   when the splicing is taking place (it requires inspection of the RTP
   payload data, rather than just the RTP headers). A splicer working as
   an RTP mixer splits the flow between the sender and receiver into
   two, and requires separate control loops, for RTCP and congestion
   control, see Section 4.4 of [RFC6828].

   A splicer implemented as an RTP translator [RFC3550] will forward the
   RTP packets from the original and substitutive senders with their
   SSRCs intact, but will need to rewrite RTCP sender report packets to
   account for the splicing. In this case, the congestion control loops
   run between original sender and receiver, and between the
   substitutive sender and receiver. The splicer needs to ensure that
   the RTCP feedback message from the receiver are passed to the right
   sender to let the congestion control work.

2.2 Overview of Splicing Interval

   To handle splicing on the RTP layer at the reserved time slots set by
   the main RTP sender, the splicer must first know the Splicing
   Interval from the main RTP sender before it can start splicing. The
   splicer can be a mixer as described in [RFC6828].

   When a new splicing is forthcoming, the main RTP sender needs to send
   the Splicing Interval to the splicer. The Splicing Interval SHOULD be
   sent by RTP header extension or RTCP extension message more than once
   to mitigate the possible packet loss. To enable the splicer to get
   the substitutive content before the splicing starts, the main RTP
   sender MUST send the Splicing Interval far ahead. For example, the
   main RTP sender can estimate when to send the Splicing Interval based
   on the round-trip time (RTT) following the mechanisms in section
   6.4.1 of [RFC3550] when the splicer sends RTCP RR to the main sender.

   The substitutive sender also needs to learn the Splicing Interval
   from the main RTP sender in advance, and thus estimates when to
   transfer the substitutive content to the splicer. The Splicing



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   Interval could be transmitted from the main RTP sender to the
   substitutive content using some out-of-band mechanisms, for example,
   a proprietary mechanism to exchange the Splicing Interval, or the
   substitutive sender is implemented together with the main RTP sender
   inside a single device. To ensure the Splicing Interval is valid for
   both the main RTP sender and the substitutive RTP sender, the two
   senders MUST share a common reference clock so that the splicer can
   achieve accurate splicing. The requirements for the common reference
   clock (e.g. resolution, skew) depend on the codec used by the media
   content.

   In this document, the main RTP sender uses a pair of NTP-format
   timestamps, to indicate when to start and end the splicing to the
   splicer: the timestamp of the first substitutive RTP packet at the
   splicing in point, and the timestamp of the first main RTP packet at
   the splicing out point.

   When the substitutive RTP sender gets the Splicing Interval, it must
   prepare the substitutive stream. The main and the substitutive
   content providers MUST ensure that the RTP timestamp of the first
   substitutive RTP packet that would be presented to the receivers
   corresponds to the same time instant as the former NTP-format
   timestamp in the Splicing Interval. To enable the splicer to know the
   first substitutive RTP packet it needs to send, the substitutive RTP
   sender MUST send the substitutive RTP packet ahead of the Splicing In
   point, allowing the splicer to find out the timestamp of this first
   RTP packet in the substitutive RTP stream, e.g., using a prior RTCP
   SR (Sender Report) message.

   When the splicing will end, the main content provider and the
   substitutive content provider MUST ensure the RTP timestamp of the
   first main RTP packet that would be presented on the receivers
   corresponds to the same time instant as the latter NTP-format
   timestamp in the Splicing Interval.


3  Conveying Splicing Interval in RTP/RTCP extensions

   This memo defines two backwards compatible RTP extensions to convey
   the Splicing Interval to the splicer: an RTP header extension and an
   RTCP splicing notification message.


3.1 RTP Header Extension


   The RTP header extension mechanism defined in [RFC5285] can be
   adapted to carry the Splicing Interval consisting of a pair of NTP-



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   format timestamps.

   This RTP header extension carries the 7 octets splicing-out NTP-
   format timestamp (lower 24-bit part of the Seconds of a NTP-format
   timestamp and the 32 bits of the Fraction of a NTP-format timestamp
   as defined in [RFC5905]), followed by the 8 octets splicing-in NTP-
   format timestamp (64-bit NTP-format timestamp as defined in
   [RFC5905]). The top 8 bits of the splicing-out NTP timestamp are
   inferred from the top 8 bits of the splicing-in NTP timestamp, under
   the assumption that the splicing-out time is after the splicing-in
   time, and the splicing interval is less than 2^25 seconds. Therefore,
   if the value of 7 octets splicing-out NTP-format timestamp is smaller
   than the value of 7 lower octets splicing-in NTP-format, it implies a
   wrap of the 56-bit splicing-out NTP-format timestamp which means the
   top 8-bit value of the 64-bit splicing-out is equal to the top 8-bit
   value of splicing-in NTP Timestamp plus 0x01. Otherwise, the top 8
   bits of splicing-out NTP timestamp is equal to the top 8 bits of
   splicing-in NTP Timestamp.

   This RTP header extension can be encoded using either the one-byte or
   two-byte header defined in [RFC5285]. Figure 1 and 2 show the
   splicing interval header extension with each of the two header
   formats.

        0                   1                   2                   3
        0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+E
      |   ID  | L=14  | OUT NTP timestamp format - Seconds (bit 8-31) |x
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+t
      |       OUT NTP timestamp format - Fraction (bit 0-31)          |e
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+n
      |        IN  NTP timestamp format - Seconds (bit 0-31)          |s
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+i
      |        IN  NTP timestamp format - Fraction (bit 0-31)         |o
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+n

                      Figure 1: Splicing Interval
                    Using the One-Byte Header Format













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        0                   1                   2                   3
        0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+E
      |   ID          |    L=15       | OUT NTP timestamp  - Seconds  |x
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+t
      |Out Secds(cont)|      OUT NTP timestamp format - Fraction      |e
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+n
      |Out Fract(cont)|      IN  NTP timestamp format - Seconds       |s
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+i
      | In Secds(cont)|     IN  NTP timestamp format - Fraction       |o
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+n
      | In Fract(cont)| 0 (pad)       |              ...
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                      Figure 2: Splicing Interval
                    Using the Two-Byte Header Format


   Since the inclusion of an RTP header extension will reduce the
   efficiency of RTP header compression, it is RECOMMENDED that the main
   sender inserts the RTP header extensions into only a number of RTP
   packets, instead of all the RTP packets, prior to the splicing in.

   After the splicer obtains the RTP header extension and derives the
   Splicing Interval, it generates its own stream and is not allowed to
   include the RTP header extension in outgoing packets to reduce header
   overhead.

3.2 RTCP Splicing Notification Message

   In addition to the RTP header extension, the main RTP sender includes
   the Splicing Interval in an RTCP splicing notification message.
   Whether or not the timestamps are included in the RTP header
   extension, the main RTP sender MUST send the RTCP splicing
   notification message. This provide robustness in the case where a
   middlebox strips RTP header extensions. The main RTP sender MUST make
   sure the splicing information contained in the RTCP splicing
   notification message consistent with the information included in the
   RTP header extensions.

   The RTCP splicing notification message is a new RTCP packet type. It
   has a fixed header followed by a pair of NTP-format timestamps:


         0                   1                   2                   3
        0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       |V=2|P|reserved |    PT=TBA   |              length             |



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       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       |                           SSRC                                |
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       |             IN NTP Timestamp (most significant word)          |
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       |             IN NTP Timestamp (least significant word)         |
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       |             OUT NTP Timestamp (most significant word)         |
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       |             OUT NTP Timestamp (least significant word)        |
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                   Figure 2: RTCP Splicing Notification Message

   The RSI packet includes the following fields:

   Length: 16 bits

      As defined in [RFC3550], the length of the RTCP packet in 32-bit
      words minus one, including the header and any padding.


   SSRC: 32 bits

      The SSRC of the Main RTP Sender.

   Timestamp: 64 bits

      Indicates the wallclock time when this splicing starts and ends.
      The full-resolution NTP-format timestamp is used, which is a 64-
      bit, unsigned, fixed-point number with the integer part in the
      first 32 bits and the fractional part in the last 32 bits.  This
      format is same as the NTP timestamp field in the RTCP Sender
      Report (Section 6.4.1 of [RFC3550]).

   The RTCP splicing notification message can be included in the RTCP
   compound packet together with RTCP SR generated at the main RTP
   sender, and hence follows the compound RTCP rules defined in Section
   6.1 in [RFC3550].


   If the use of non-compound RTCP [RFC5506] was previously negotiated
   between the sender and the splicer, the RTCP splicing notification
   message may be sent as non-compound RTCP packets. In some cases that
   the mapping from RTP timestamp to NTP timestamp changes, e.g., clock
   drift happening before the splicing event, it may be required to send
   RTCP SR or even updated Splicing Interval information timely to
   update the timestamp mapping for accurate splicing.



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   Since the RTCP splicing notification message is intentionally sent by
   the main RTP sender to the splicer, the splicer is not allowed to
   forward this message to the receivers so as to avoid their useless
   processing and additional RTCP bandwidth consumption in the
   downstream.

4  Reducing Splicing Latency

   When splicing starts or ends, the splicer outputs the multimedia
   content from another sender to the receivers. Given that the
   receivers must first acquire certain information ([RFC6285] refers to
   this information as Reference Information) to start processing the
   multimedia data, either the main RTP sender or the substitutive
   sender SHOULD provide the Reference Information together with its
   multimedia content to reduce the delay caused by acquiring the
   Reference Information. The methods by which the Reference Information
   is distributed to the receivers is out of scope of this memo.

   Another latency element is synchronization caused delay. The
   receivers must receive enough synchronization metadata prior to
   synchronizing the separate components of the multimedia streams when
   splicing starts or ends. Either the main RTP sender or the
   substitutive sender SHOULD send the synchronization metadata early
   enough so that the receivers can play out the multimedia in a
   synchronized fashion. The main RTP sender or the substitutive sender
   can estimate when to send the synchronization metadata based on, for
   example, the round-trip time (RTT) following the mechanisms in
   section 6.4.1 of [RFC3550] when the splicer sends RTCP RR to the main
   sender or the substitutive sender. The main RTP sender and the
   substitutive sender can also be coordinated by some proprietary out-
   of-band mechanisms to decide when and whom to send the metadata. If
   both send the information, the splicer SHOULD pick one based on the
   current situation, e.g., choosing main RTP sender when synchronizing
   the main media content while choosing the information from the
   substitutive sender when synchronizing the spliced content. The
   mechanisms defined in [RFC6051] are RECOMMENDED to be adopted to
   reduce the possible synchronization delay.














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5  Failure Cases

   This section examines the implications of losing RTCP splicing
   notification message and the other failure case, e.g., the RTP header
   extension is stripped on the path.

   Given that there may be a splicing un-aware middlebox on the path
   between the main RTP sender and the splicer, the main and the
   substitutive RTP senders can use one heuristic to verify whether or
   not the Splicing Interval reaches the splicer.

   The splicer can be implemented to have its own SSRC, and send RTCP
   reception reports to the senders of the main and substitutive RTP
   streams. This allows the senders to detect problems on the path to
   the splicer. Alternatively, it is possible to implement the splicer
   such that it has no SSRC, and does not send RTCP reports; this
   prevents the senders from being able to monitor the quality to the
   path to the splicer.

   If the splicer has an SSRC and sends its own RTCP reports, it can
   choose not to pass RTCP reports it receives from the receivers to the
   senders. This will stop the senders from being able to monitor the
   quality of the paths from the splicer to the receivers.

   A splicer that has an SSRC can choose to pass RTCP reception reports
   from the receivers back to the senders, after modifications to
   account for the splicing. This will allow the senders the monitor the
   quality of the paths from the splicer to the receivers. A splicer
   that does not have its own SSRC has to forward and translation RTCP
   reports from the receiver, otherwise the senders will not see any
   receivers in the RTP session.

   If the splicer is implemented following [RFC6828], it will have its
   own SSRC and will send its own RTCP reports, and will forward
   translated RTCP reports from the receivers.

   Upon the detection of a failure, the splicer can communicate with the
   main sender and the substitutive sender in some out of band signaling
   ways to fall back to the payload specific mechanisms it supports,
   e.g., MPEG-TS splicing solution defined in [SCTE35], or just abandon
   the splicing.

6 Session Description Protocol (SDP) Signaling

   This document defines the URI for declaring this header extension in
   an extmap attribute to be "urn:ietf:params:rtp-hdrext:splicing-
   interval".




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   This document extends the standard semantics defined in SDP Grouping
   Framework [RFC5888] with a new semantic: SPLICE to represent the
   relationship between the main RTP stream and the substitutive RTP
   stream. Only 2 m-lines are allowed in the SPLICE group. The main RTP
   stream is the one with the extended extmap attribute, and the other
   one is substitutive stream. A single m-line MUST NOT be included in
   different SPLICE groups at the same time. The main RTP sender
   provides the information about both main and substitutive sources.

   The extended SDP attribute specified in this document is applicable
   for offer/answer content [RFC3264] and do not affect any rules when
   negotiating offer and answer. When used with multiple m-lines,
   substitutive RTP MUST be applied only to the RTP packets whose SDP m-
   line is in the same group with the substitutive stream using SPLICE
   and has the extended splicing extmap attribute. This semantic is also
   applicable for BUNDLE cases.

   The following examples show how SDP signaling could be used for
   splicing in different cases.

6.1 Declarative SDP

              v=0
              o=xia 1122334455 1122334466 IN IP4 splicing.example.com
              s=RTP Splicing Example
              t=0 0
              a=group:SPLICE 1 2
              m=video 30000 RTP/AVP 100
              i=Main RTP Stream
              c=IN IP4 233.252.0.1/127
              a=rtpmap:100 MP2T/90000
              a=extmap:1 urn:ietf:params:rtp-hdrext:splicing-interval
              a=mid:1
              m=video 30002 RTP/AVP 100
              i=Substitutive RTP Stream
              c=IN IP4 233.252.0.2/127
              a=sendonly
              a=rtpmap:100 MP2T/90000
              a=mid:2

         Figure 3: Example SDP for a single-channel splicing scenario

   The splicer receiving the SDP message above receives one MPEG2-TS
   stream (payload 100) from the main RTP sender (with multicast
   destination address of 233.252.0.1) on port 30000, and/or receives
   another MPEG2-TS stream from the substitutive RTP sender (with
   multicast destination address of 233.252.0.2) on port 30002.  But at
   a particular point in time, the splicer only selects one stream and



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   outputs the content from the chosen stream to the downstream
   receivers.

6.2 Offer/Answer without BUNDLE

   SDP Offer - from main RTP sender

            v=0
            o=xia 1122334455 1122334466 IN IP4 splicing.example.com
            s=RTP Splicing Example
            t=0 0
            a=group:SPLICE 1 2
            m=video 30000 RTP/AVP 31 100
            i=Main RTP Stream
            c=IN IP4 splicing.example.com
            a=rtpmap:31 H261/90000
            a=rtpmap:100 MP2T/90000
            a=extmap:1 urn:ietf:params:rtp-hdrext:splicing-interval
            a=sendonly
            a=mid:1
            m=video 40000 RTP/AVP 31 100
            i=Substitutive RTP Stream
            c=IN IP4 substitutive.example.com
            a=rtpmap:31 H261/90000
            a=rtpmap:100 MP2T/90000
            a=sendonly
            a=mid:2

   SDP Answer - from splicer

            v=0
            o=xia 1122334455 1122334466 IN IP4 splicer.example.com
            s=RTP Splicing Example
            t=0 0
            a=group:SPLICE 1 2
            m=video 30000 RTP/AVP 100
            i=Main RTP Stream
            c=IN IP4 splicer.example.com
            a=rtpmap:100 MP2T/90000
            a=extmap:1 urn:ietf:params:rtp-hdrext:splicing-interval
            a=recvonly
            a=mid:1
            m=video 40000 RTP/AVP 100
            i=Substitutive RTP Stream
            c=IN IP4 splicer.example.com
            a=rtpmap:100 MP2T/90000
            a=recvonly
            a=mid:2



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6.3 Offer/Answer with BUNDLE: All Media are spliced

   In this example, the bundled audio and video media have their own
   substitutive media for splicing:

   1.  An Offer, in which the offerer assigns a unique address and a
   substitutive media to each bundled "m="line for splicing within the
   BUNDLE group.

   2.  An answer, in which the answerer selects its own BUNDLE address,
   and leave the substitutive media untouched.

   SDP Offer - from main RTP sender


          v=0
          o=alice 1122334455 1122334466 IN IP4 splicing.example.com
          s=RTP Splicing Example
          c=IN IP4 splicing.example.com
          t=0 0
          a=group:SPLICE foo 1
          a=group:SPLICE bar 2
          a=group:BUNDLE foo bar
          m=audio 10000 RTP/AVP 0 8 97
          a=mid:foo
          b=AS:200
          a=rtpmap:0 PCMU/8000
          a=rtpmap:8 PCMA/8000
          a=rtpmap:97 iLBC/8000
          a=extmap:1 urn:ietf:params:rtp-hdrext:splicing-interval
          a=sendonly
          m=video 10002 RTP/AVP 31 32
          a=mid:bar
          b=AS:1000
          a=rtpmap:31 H261/90000
          a=rtpmap:32 MPV/90000
          a=extmap:2 urn:ietf:params:rtp-hdrext:splicing-interval
          a=sendonly
          m=audio 20000 RTP/AVP 0 8 97
          i=Substitutive audio RTP Stream
          c=IN IP4 substitive.example.com
          a=rtpmap:0 PCMU/8000
          a=rtpmap:8 PCMA/8000
          a=rtpmap:97 iLBC/8000
          a=sendonly
          a=mid:1
          m=video 20002 RTP/AVP 31 32
          i=Substitutive video RTP Stream



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          c=IN IP4 substitive.example.com
          a=rtpmap:31 H261/90000
          a=rtpmap:32 MPV/90000
          a=mid:2
          a=sendonly

   SDP Answer - from the splicer

          v=0
          o=bob 2808844564 2808844564 IN IP4 splicer.example.com
          s=RTP Splicing Example
          c=IN IP4 splicer.example.com
          t=0 0
          a=group:SPLICE foo 1
          a=group:SPLICE bar 2
          a=group:BUNDLE foo bar
          m=audio 30000 RTP/AVP 0
          a=mid:foo
          b=AS:200
          a=rtpmap:0 PCMU/8000
          a=extmap:1 urn:ietf:params:rtp-hdrext:splicing-interval
          a=recvonly
          m=video 30000 RTP/AVP 32
          a=mid:bar
          b=AS:1000
          a=rtpmap:32 MPV/90000
          a=extmap:2 urn:ietf:params:rtp-hdrext:splicing-interval
          a=recvonly
          m=audio 30002 RTP/AVP 0
          i=Substitutive audio RTP Stream
          c=IN IP4 splicer.example.com
          a=rtpmap:0 PCMU/8000
          a=recvonly
          a=mid:1
          m=video 30004 RTP/AVP 32
          i=Substitutive video RTP Stream
          c=IN IP4 splicer.example.com
          a=rtpmap:32 MPV/90000
          a=mid:2
          a=recvonly

6.4 Offer/Answer with BUNDLE: a Subset of Media are Spliced

   In this example, the substitutive media only applies for video when
   splicing:

   1. An Offer, in which the offerer assigns a unique address to each
   bundled "m="line within the BUNDLE group, and assigns a substitutive



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   media to the bundled video "m=" line for splicing.

   2. An answer, in which the answerer selects its own BUNDLE address,
   and leave the substitutive media untouched.

   SDP Offer - from the main RTP sender:

          v=0
          o=alice 1122334455 1122334466 IN IP4 splicing.example.com
          s=RTP Splicing Example
          c=IN IP4 splicing.example.com
          t=0 0
          a=group:SPLICE bar 2
          a=group:BUNDLE foo bar
          m=audio 10000 RTP/AVP 0 8 97
          a=mid:foo
          b=AS:200
          a=rtpmap:0 PCMU/8000
          a=rtpmap:8 PCMA/8000
          a=rtpmap:97 iLBC/8000
          a=sendonly
          m=video 10002 RTP/AVP 31 32
          a=mid:bar
          b=AS:1000
          a=rtpmap:31 H261/90000
          a=rtpmap:32 MPV/90000
          a=extmap:2 urn:ietf:params:rtp-hdrext:splicing-interval
          a=sendonly
          m=video 20000 RTP/AVP 31 32
          i=Substitutive video RTP Stream
          c=IN IP4 substitutive.example.com
          a=rtpmap:31 H261/90000
          a=rtpmap:32 MPV/90000
          a=mid:2
          a=sendonly

   SDP Answer - from the splicer:

          v=0
          o=bob 2808844564 2808844564 IN IP4 splicer.example.com
          s=RTP Splicing Example
          c=IN IP4 splicer.example.com
          t=0 0
          a=group:SPLICE bar 2
          a=group:BUNDLE foo bar
          m=audio 30000 RTP/AVP 0
          a=mid:foo
          b=AS:200



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          a=rtpmap:0 PCMU/8000
          a=recvonly
          m=video 30000 RTP/AVP 32
          a=mid:bar
          b=AS:1000
          a=rtpmap:32 MPV/90000
          a=extmap:2 urn:ietf:params:rtp-hdrext:splicing-interval
          a=recvonly
          m=video 30004 RTP/AVP 32
          i=Substitutive video RTP Stream
          c=IN IP4 splicer.example.com
          a=rtpmap:32 MPV/90000
          a=mid:2
          a=recvonly

7  Security Considerations

   The security considerations of the RTP specification [RFC3550] and
   the general mechanism for RTP header extensions [RFC5285] apply. The
   splicer can either be a mixer or a translator, and all the security
   considerations of these two RTP intermediaries topologies described
   in [RFC7667] and [RFC7201] are applicable for the splicer.

   The splicer replaces some content with other content in RTP packet,
   thus breaking any RTP-level end-to-end security, such as source
   authentication and integrity protection. End to end source
   authentication is not possible with any known existing splicing
   solution. A new solution can theoretically be developed that enables
   identification of the participating entities and what each provides,
   i.e., the different media sources, main and substituting, and the
   splicer which provides the RTP-level integration of the media
   payloads in a common timeline and synchronization context.

   Since the splicer breaks RTP-level end-to-end security, it needs to
   be part of the signaling context and the necessary security
   associations (e.g., SRTP crypto contexts) established for the RTP
   session participants. When using the Secure Real-Time Transport
   Protocol (SRTP) [RFC3711], the splicer would have to be provisioned
   with the same security association as the main RTP sender.

   If there is a concern about the confidentiality of the splicing time
   information, the header extension defined in this document MUST be
   also protected, for example, header extension encryption [RFC6904]
   can be used in this case. However, the malicious endpoint may get the
   splicing time information by other means, e.g., inferring from the
   communication between the main and substitutive content sources. To
   avoid the insertion of invalid substitutive content, the splicer MUST
   have some mechanisms to authenticate the substitutive stream source.



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   For cases that the splicing time information is changed by a
   malicious endpoint, the splicing, for example, may fail since it will
   not be available at the right time for the substitutive media to
   arrive. Another case is that an attacker may prevent the receivers
   receiving the content from the main sender by inserting extra
   splicing time information. To avoid the above cases happening, the
   authentication of the RTP header extension for splicing time
   information SHOULD be considered.

   When a splicer implemented as a mixer sends the stream to the
   receivers, CSRC list, which can be used to detect RTP-level
   forwarding loops as defined in Section 8.2 of [RFC3550], may be
   removed for simplifying the receivers that can not handle multiple
   sources in the RTP stream. Hence, loops may occur to cause packets to
   loop back to upstream of the splicer and may form a serious denial-
   of-service threat. In such a case, non-RTP means, e.g., signaling
   among all the participants, MUST be used to detect and resolve loops.

8  IANA Considerations

8.1  RTCP Control Packet Types

   Based on the guidelines suggested in [RFC5226], a new RTCP packet
   format has been registered with the RTCP Control Packet Type (PT)
   Registry:

         Name: SNM

         Long name: Splicing Notification Message

         Value: TBA

         Reference: This document

8.2  RTP Compact Header Extensions

   The IANA has also registered a new RTP Compact Header Extension
   [RFC5285], according to the following:

         Extension URI: urn:ietf:params:rtp-hdrext:splicing-interval

         Description: Splicing Interval

         Contact: Jinwei Xia <xiajinwei@huawei.com>

         Reference: This document

8.3 SDP Grouping Semantic Extension



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   This document request IANA to register the new SDP grouping semantic
   extension called "SPLICE".

         Semantics: Splice

         Token:SPLICE


         Reference: This document

9 Acknowledgement

   The authors would like to thank the following individuals who help to
   review this document and provide very valuable comments: Colin
   Perkins, Bo Burman, Stephen Botzko, Ben Campbell.

10  References

10.1  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.

   [RFC3264]  Rosenberg, J., and H. Schulzrinne, "An Offer/Answer Model
              with the Session Description Protocol (SDP)", RFC 3264,
              June 2002.

   [RFC5285]  Singer, D. and H. Desineni, "A General Mechanism for RTP
              Header Extensions", RFC 5285, July 2008.

   [RFC5888]  Camarillo, G. and H. Schulzrinne, "The Session Description
              Protocol (SDP) Grouping Framework", RFC 5888, June 2010.

   [RFC5905]  Mills, D., Martin, J., Ed., Burbank, J., and W. Kasch,
              "Network Time Protocol Version 4: Protocol and Algorithms
              Specification", RFC 5905, June 2010.

   [RFC6051]  Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
              Flows", RFC 6051, November 2010.

   [RFC7201]  Westerlund, M. and C. Perkins, "Options for Securing RTP
              Sessions", RFC 7201, April 2014.

   [RFC7667]  Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,



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              November 2015.

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.


   [RFC5226]  Narten, T. and H. Alvestrand, "Guidelines for Writing an
              IANA Considerations Section in RFCs", BCP 26, RFC 5226,
              May 2008.

   [RFC5506]  Johansson, I. and M. Westerlund, "Support for Reduced-Size
              Real-Time Transport Control Protocol (RTCP): Opportunities
              and Consequences", RFC 5506, April 2009.

   [RFC6285]  Ver Steeg, B., Begen, A., Van Caenegem, T., and Z. Vax,
              "Unicast-Based Rapid Acquisition of Multicast RTP
              Sessions", RFC 6285, June 2011.

   [RFC6904]  Lennox, J.,"Encryption of Header Extensions in the Secure
              Real-Time Transport Protocol (SRTP)", April 2013.

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

   [RFC6828]  Xia, J., "Content Splicing for RTP Sessions", RFC 6828,
              January 2013.


Authors' Addresses


   Jinwei Xia
   Huawei

   Email: xiajinwei@huawei.com

   Roni Even
   Huawei

   Email: ron.even.tlv@gmail.com


   Rachel Huang
   Huawei



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   Email: rachel.huang@huawei.com


   Lingli Deng
   China Mobile

   Email: denglingli@chinamobile.com












































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