<pre>Internet Engineering Task Force (IETF)                            J. Xia
Request for Comments: 8286                                       R. Even
Category: Standards Track                                       R. Huang
ISSN: 2070-1721                                                   Huawei
                                                                 L. Deng
                                                            China Mobile
                                                            October 2017


            <span class="h1">RTP/RTCP Extension for RTP Splicing Notification</span>

Abstract

   Content splicing is a process that replaces the content of a main
   multimedia stream with other multimedia content and that 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 RTP Control Protocol (RTCP)
   packet that conveys the information out of band.

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 <a href="./rfc7841#section-2">Section&nbsp;2 of RFC 7841</a>.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at
   <a href="https://www.rfc-editor.org/info/rfc8286">https://www.rfc-editor.org/info/rfc8286</a>.














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Copyright Notice

   Copyright (c) 2017 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to <a href="https://www.rfc-editor.org/bcp/bcp78">BCP 78</a> and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (<a href="https://trustee.ietf.org/license-info">https://trustee.ietf.org/license-info</a>) 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

   <a href="#section-1">1</a>. Introduction ....................................................<a href="#page-3">3</a>
      <a href="#section-1.1">1.1</a>. Terminology ................................................<a href="#page-3">3</a>
   <a href="#section-2">2</a>. Overview ........................................................<a href="#page-4">4</a>
      <a href="#section-2.1">2.1</a>. Overview of RTP Splicing ...................................<a href="#page-4">4</a>
      <a href="#section-2.2">2.2</a>. Overview of Splicing Interval ..............................<a href="#page-5">5</a>
   <a href="#section-3">3</a>. Conveying Splicing Interval in RTP/RTCP Extensions ..............<a href="#page-7">7</a>
      <a href="#section-3.1">3.1</a>. RTP Header Extension .......................................<a href="#page-7">7</a>
      <a href="#section-3.2">3.2</a>. RTCP Splicing Notification Message .........................<a href="#page-8">8</a>
   <a href="#section-4">4</a>. Reducing Splicing Latency ......................................<a href="#page-10">10</a>
   <a href="#section-5">5</a>. Failure Cases ..................................................<a href="#page-11">11</a>
   <a href="#section-6">6</a>. Session Description Protocol (SDP) Signaling ...................<a href="#page-12">12</a>
      <a href="#section-6.1">6.1</a>. Declarative SDP ...........................................<a href="#page-12">12</a>
      <a href="#section-6.2">6.2</a>. Offer/Answer without BUNDLE ...............................<a href="#page-13">13</a>
      <a href="#section-6.3">6.3</a>. Offer/Answer with BUNDLE: All Media Are Spliced ...........<a href="#page-14">14</a>
      <a href="#section-6.4">6.4</a>. Offer/Answer with BUNDLE: A Subset of Media Are Spliced ...<a href="#page-16">16</a>
   <a href="#section-7">7</a>. Security Considerations ........................................<a href="#page-18">18</a>
   <a href="#section-8">8</a>. IANA Considerations ............................................<a href="#page-19">19</a>
      <a href="#section-8.1">8.1</a>. RTCP Control Packet Types .................................<a href="#page-19">19</a>
      <a href="#section-8.2">8.2</a>. RTP Compact Header Extensions .............................<a href="#page-20">20</a>
      <a href="#section-8.3">8.3</a>. SDP Grouping Semantic Extension ...........................<a href="#page-20">20</a>
   <a href="#section-9">9</a>. References .....................................................<a href="#page-20">20</a>
      <a href="#section-9.1">9.1</a>. Normative References ......................................<a href="#page-20">20</a>
      <a href="#section-9.2">9.2</a>. Informative References ....................................<a href="#page-21">21</a>
   Acknowledgements ..................................................<a href="#page-22">22</a>
   Authors' Addresses ................................................<a href="#page-22">22</a>









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<span class="h2"><a class="selflink" id="section-1" href="#section-1">1</a>.  Introduction</span>

   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".

   [<a id="ref-SCTE35">SCTE35</a>] provides a method that encapsulates the Splicing Interval
   inside the MPEG2-TS (MPEG2 transport stream) layer in cable TV
   systems.  When transported in RTP, a middlebox 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 middlebox and limits the
   number of RTP sessions the middlebox can support.

   This document defines an RTP header extension [<a href="./rfc8285" title="&quot;A General Mechanism for RTP Header Extensions&quot;">RFC8285</a>] 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-unaware
   middlebox on the path between the RTP sender and the splicer might
   strip this RTP header extension.

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

<span class="h3"><a class="selflink" id="section-1.1" href="#section-1.1">1.1</a>.  Terminology</span>

   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
   <a href="https://www.rfc-editor.org/bcp/bcp14">BCP 14</a> [<a href="./rfc2119" title="&quot;Key words for use in RFCs to Indicate Requirement Levels&quot;">RFC2119</a>] [<a href="./rfc8174" title="&quot;Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words&quot;">RFC8174</a>] when, and only when, they appear in all
   capitals, as shown here.










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   In addition, we define the following terms:

   Main RTP Sender:

      The sender of RTP packets carrying the main RTP stream.

   Splicer:

      An intermediary node that inserts substitutive content into a main
      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 timestamps, representing the main RTP sender wallclock
      time, for the splicing-in point and splicing-out point per
      [<a href="./rfc6828" title="&quot;Content Splicing for RTP Sessions&quot;">RFC6828</a>], 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.

<span class="h2"><a class="selflink" id="section-2" href="#section-2">2</a>.  Overview</span>

<span class="h3"><a class="selflink" id="section-2.1" href="#section-2.1">2.1</a>.  Overview of RTP Splicing</span>

   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




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   scenario is where an 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 the
   main content and substitutive content simultaneously but only chooses
   to send one of them to the receiver at any point in 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.
   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 an RTP translator.  If implemented as an RTP mixer, the
   splicer will use its own synchronization source (SSRC), sequence
   number space, and timing model when generating the output stream to
   receivers, using the contributing source (CSRC) list to indicate
   whether the original content 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 it requires
   separate control loops for RTCP and congestion control.  [<a href="./rfc6828" title="&quot;Content Splicing for RTP Sessions&quot;">RFC6828</a>]
   provides an example of an RTP mixer approach.

   A splicer implemented as an RTP translator [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>] will forward the
   RTP packets from the original and substitutive senders with their
   SSRCs intact but will need to rewrite RTCP Sender Report (SR) packets
   to account for the splicing.  In this case, the congestion control
   loops run between the original sender and receiver and between the
   substitutive sender and receiver.  The splicer needs to ensure that
   the RTCP feedback messages from the receiver are passed to the right
   sender to let the congestion control work.

<span class="h3"><a class="selflink" id="section-2.2" href="#section-2.2">2.2</a>.  Overview of Splicing Interval</span>

   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.

   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 the RTP header extension or RTCP extension message more



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   than once to mitigate 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 well in advance.  For example,
   the main RTP sender can estimate when to send the Splicing Interval
   based on the round-trip time (RTT), following the mechanisms
   described in <a href="./rfc3550#section-6.4.1">Section&nbsp;6.4.1 of [RFC3550]</a> when the splicer sends an
   RTCP Receiver Report (RR) to the main sender.

   The substitutive sender also needs to learn the Splicing Interval
   from the main RTP sender in advance and estimate when to transfer the
   substitutive content to the splicer.  The Splicing 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 that 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 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 content provider and the
   substitutive content provider 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 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 message.

   When it is time for the splicing to end, the main content provider
   and the substitutive content provider MUST ensure that 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 timestamp in the Splicing Interval.






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<span class="h2"><a class="selflink" id="section-3" href="#section-3">3</a>.  Conveying Splicing Interval in RTP/RTCP Extensions</span>

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

<span class="h3"><a class="selflink" id="section-3.1" href="#section-3.1">3.1</a>.  RTP Header Extension</span>

   The RTP header extension mechanism defined in [<a href="./rfc8285" title="&quot;A General Mechanism for RTP Header Extensions&quot;">RFC8285</a>] can be
   adapted to carry the Splicing Interval, which consists of a pair of
   NTP timestamps.

   This RTP header extension carries the 7 octets of the splicing-out
   NTP timestamp (lower 24-bit part of the "Seconds" of an NTP timestamp
   and the 32 bits of the "Fraction" of an NTP timestamp as defined in
   [<a href="./rfc5905" title="&quot;Network Time Protocol Version 4: Protocol and Algorithms Specification&quot;">RFC5905</a>]), followed by the 8 octets of the splicing-in NTP timestamp
   (64-bit NTP timestamp as defined in [<a href="./rfc5905" title="&quot;Network Time Protocol Version 4: Protocol and Algorithms Specification&quot;">RFC5905</a>]).  The top 8 bits of
   the splicing-out NTP timestamp are inferred from the top 8 bits of
   the splicing-in NTP timestamp, assuming that (1) the splicing-out
   time is after the splicing-in time and (2) the Splicing Interval is
   less than 2^25 seconds.  Therefore, if the value of the 7 octets of
   the splicing-out NTP timestamp is smaller than the value of the
   7 lower octets of the splicing-in NTP timestamp, it implies a wrap of
   the 56-bit splicing-out NTP timestamp, which means that the top 8-bit
   value of the 64-bit splicing-out NTP timestamp is equal to the top
   8-bit value of the splicing-in NTP timestamp plus 0x01.  Otherwise,
   the top 8 bits of the splicing-out NTP timestamp are equal to the top
   8 bits of the splicing-in NTP timestamp.

   This RTP header extension can be encoded using either the one-byte or
   two-byte header defined in [<a href="./rfc8285" title="&quot;A General Mechanism for RTP Header Extensions&quot;">RFC8285</a>].  Figures 1 and 2 show the
   Splicing Interval header extension with each of the two header
   formats.


















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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=14  |    OUT NTP timestamp - Seconds (bit 8-31)     |x
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+t
   |          OUT NTP timestamp - Fraction (bit 0-31)              |e
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+n
   |           IN NTP timestamp - Seconds (bit 0-31)               |s
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+i
   |           IN NTP timestamp - Fraction (bit 0-31)              |o
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+n

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

    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 - Fraction          |e
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+n
   |OUT Fract(cont)|          IN NTP timestamp - Seconds           |s
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+i
   | IN Secds(cont)|          IN NTP timestamp - 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 insert the RTP header extensions into a number of RTP packets,
   instead of all of 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; this reduces
   header overhead.

<span class="h3"><a class="selflink" id="section-3.2" href="#section-3.2">3.2</a>.  RTCP Splicing Notification Message</span>

   In addition to including 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 provides robustness in the case
   where a middlebox strips RTP header extensions.  The main RTP sender



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   MUST make sure that the splicing information contained in the RTCP
   splicing notification message is 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 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=213   |              length             |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                           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 3: RTCP Splicing Notification Message

   The RTCP splicing notification message includes the following fields:

   Length: 16 bits

      As defined in [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>], 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 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 the same as the NTP timestamp field in the RTCP SR
      (<a href="./rfc3550#section-6.4.1">Section&nbsp;6.4.1 of [RFC3550]</a>).







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   The RTCP splicing notification message can be included in the RTCP
   compound packet together with the RTCP SR generated at the main RTP
   sender; hence, it follows the compound RTCP rules defined in
   <a href="./rfc3550#section-6.1">Section&nbsp;6.1 in [RFC3550]</a>.

   If the use of non-compound RTCP [<a href="./rfc5506" title="&quot;Support for Reduced-Size Real-Time Transport Control Protocol (RTCP): Opportunities and Consequences&quot;">RFC5506</a>] was previously negotiated
   between the sender and the splicer, the RTCP splicing notification
   messages may be sent as non-compound RTCP packets.  In some cases
   where the mapping from the RTP timestamp to the NTP timestamp
   changes, e.g., clock drift happens before the splicing event, sending
   an RTCP SR or even updated Splicing Interval information in a timely
   manner might be required in order to update the timestamp mapping for
   accurate splicing.

   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 useless
   processing and additional RTCP bandwidth consumption in the
   downstream receivers.

<span class="h2"><a class="selflink" id="section-4" href="#section-4">4</a>.  Reducing Splicing Latency</span>

   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 ([<a href="./rfc6285" title="&quot;Unicast-Based Rapid Acquisition of Multicast RTP Sessions&quot;">RFC6285</a>] 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 are out of scope for
   this memo.

   Another latency element is delay caused by synchronization.  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 RTT, following the mechanisms described in <a href="./rfc3550#section-6.4.1">Section&nbsp;6.4.1
   of [RFC3550]</a> when the splicer sends an 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 to whom, the metadata is to be sent.
   If both send the information, the splicer SHOULD pick one based on
   the current situation, e.g., choosing either (1) the main RTP sender



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   when synchronizing the main media content or (2) the information from
   the substitutive sender when synchronizing the spliced content.  To
   reduce possible synchronization delay, it is RECOMMENDED that the
   mechanisms defined in [<a href="./rfc6051" title="&quot;Rapid Synchronisation of RTP Flows&quot;">RFC6051</a>] be adopted.

<span class="h2"><a class="selflink" id="section-5" href="#section-5">5</a>.  Failure Cases</span>

   This section examines the implications of losing RTCP splicing
   notification messages, e.g., the RTP header extension is stripped on
   the path.

   Given that there may be a splicing-unaware middlebox on the path
   between the main RTP sender and the splicer, the main and
   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 of 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 prevent 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 to 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 translate RTCP
   reports from the receiver; otherwise, the senders will not see any
   receivers in the RTP session.

   If the splicer is implemented as a mixer, it will have its own SSRC,
   send its own RTCP reports, and 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 via some out-of-band
   signaling technique and fall back to the payload-specific mechanisms
   it supports, e.g., the MPEG2-TS splicing solution defined in
   [<a href="#ref-SCTE35" title="&quot;Digital Program Insertion Cueing Message for Cable&quot;">SCTE35</a>], or just abandon the splicing.





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<span class="h2"><a class="selflink" id="section-6" href="#section-6">6</a>.  Session Description Protocol (SDP) Signaling</span>

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

   This document extends the standard semantics defined in "The Session
   Description Protocol (SDP) Grouping Framework" [<a href="./rfc5888" title="&quot;The Session Description Protocol (SDP) Grouping Framework&quot;">RFC5888</a>] with a new
   semantic, called "SPLICE", to represent the relationship between the
   main RTP stream and the substitutive RTP stream.  Only two "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 the
   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 [<a href="./rfc3264" title="&quot;An Offer/Answer Model with Session Description Protocol (SDP)&quot;">RFC3264</a>] and does not affect any rules when
   negotiating offers and answers.  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.

<span class="h3"><a class="selflink" id="section-6.1" href="#section-6.1">6.1</a>.  Declarative SDP</span>

      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 4: Example SDP for a Single-Channel Splicing Scenario



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   The splicer receiving the SDP message above receives one MPEG2-TS
   stream (payload 100) from the main RTP sender (with a multicast
   destination address of 233.252.0.1) on port 30000 and/or receives
   another MPEG2-TS stream from the substitutive RTP sender (with a
   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
   outputs the content from the chosen stream to the downstream
   receivers.

<span class="h3"><a class="selflink" id="section-6.2" href="#section-6.2">6.2</a>.  Offer/Answer without BUNDLE</span>

   SDP Offer - from the 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


















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   SDP Answer - from the 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

<span class="h3"><a class="selflink" id="section-6.3" href="#section-6.3">6.3</a>.  Offer/Answer with BUNDLE: All Media Are Spliced</span>

   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 leaves the substitutive media untouched.



















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











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   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

<span class="h3"><a class="selflink" id="section-6.4" href="#section-6.4">6.4</a>.  Offer/Answer with BUNDLE: A Subset of Media Are Spliced</span>

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

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





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   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





















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   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
      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

<span class="h2"><a class="selflink" id="section-7" href="#section-7">7</a>.  Security Considerations</span>

   The security considerations of the RTP specification [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>] and
   the general mechanism for RTP header extensions [<a href="./rfc8285" title="&quot;A General Mechanism for RTP Header Extensions&quot;">RFC8285</a>] apply.  The
   splicer can be either a mixer or a translator, and all the security
   considerations of topologies [<a href="./rfc7667" title="&quot;RTP Topologies&quot;">RFC7667</a>] [<a href="./rfc7201" title="&quot;Options for Securing RTP Sessions&quot;">RFC7201</a>] for these two types
   of RTP intermediaries are applicable for the splicer.

   The splicer replaces some content with other content in RTP packets,
   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 substitutive -- 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., Secure Real-time Transport Protocol (SRTP)



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   [<a id="ref-RFC3711">RFC3711</a>] crypto contexts) established for the RTP session
   participants.  When using SRTP, the splicer would have to be
   provisioned with the same security association as the main RTP
   sender.

   If there are concerns about the confidentiality of the splicing time
   information, the header extension defined in this document MUST also
   be protected; for example, header extension encryption [<a href="./rfc6904" title="&quot;Encryption of Header Extensions in the Secure Real-time Transport Protocol (SRTP)&quot;">RFC6904</a>] can
   be used in this case.  However, the malicious endpoint may get the
   splicing time information by other means, e.g., inferring it 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.

   For cases where 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 one where an attacker may prevent the
   receivers from receiving the content from the main sender by
   inserting extra splicing time information.  To avoid the above
   scenarios, 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, the CSRC list, which can be used to detect RTP-level
   forwarding loops as defined in <a href="./rfc3550#section-8.2">Section&nbsp;8.2 of [RFC3550]</a>, may be
   removed for simplifying the receivers that cannot handle multiple
   sources in the RTP stream.  Hence, loops may occur, causing packets
   to loop back to a point upstream of the splicer and possibly forming
   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.

<span class="h2"><a class="selflink" id="section-8" href="#section-8">8</a>.  IANA Considerations</span>

<span class="h3"><a class="selflink" id="section-8.1" href="#section-8.1">8.1</a>.  RTCP Control Packet Types</span>

   Based on the guidelines suggested in [<a href="./rfc8126" title="">RFC8126</a>], a new RTCP packet
   format has been registered in the "RTCP Control Packet types (PT)"
   registry:

      Name: SNM

      Long name: Splicing Notification Message

      Value: 213

      Reference: This document



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<span class="h3"><a class="selflink" id="section-8.2" href="#section-8.2">8.2</a>.  RTP Compact Header Extensions</span>

   IANA has registered a new RTP Compact Header Extension [<a href="./rfc8285" title="&quot;A General Mechanism for RTP Header Extensions&quot;">RFC8285</a>],
   according to the following:

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

      Description: Splicing Interval

      Contact: Jinwei Xia &lt;xiajinwei@huawei.com&gt;

      Reference: This document

<span class="h3"><a class="selflink" id="section-8.3" href="#section-8.3">8.3</a>.  SDP Grouping Semantic Extension</span>

   IANA has registered the new SDP grouping semantic extension called
   "SPLICE" in the "Semantics for the 'group' SDP Attribute" subregistry
   of the "Session Description Protocol (SDP) Parameters" registry:

      Semantics: Splice

      Token: SPLICE

      Reference: This document

<span class="h2"><a class="selflink" id="section-9" href="#section-9">9</a>.  References</span>

<span class="h3"><a class="selflink" id="section-9.1" href="#section-9.1">9.1</a>.  Normative References</span>

   [<a id="ref-RFC2119">RFC2119</a>]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", <a href="https://www.rfc-editor.org/bcp/bcp14">BCP 14</a>, <a href="./rfc2119">RFC 2119</a>,
              DOI 10.17487/RFC2119, March 1997,
              &lt;<a href="https://www.rfc-editor.org/info/rfc2119">https://www.rfc-editor.org/info/rfc2119</a>&gt;.

   [<a id="ref-RFC3264">RFC3264</a>]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", <a href="./rfc3264">RFC 3264</a>,
              DOI 10.17487/RFC3264, June 2002,
              &lt;<a href="https://www.rfc-editor.org/info/rfc3264">https://www.rfc-editor.org/info/rfc3264</a>&gt;.

   [<a id="ref-RFC3550">RFC3550</a>]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, <a href="./rfc3550">RFC 3550</a>, DOI 10.17487/RFC3550,
              July 2003, &lt;<a href="https://www.rfc-editor.org/info/rfc3550">https://www.rfc-editor.org/info/rfc3550</a>&gt;.

   [<a id="ref-RFC5888">RFC5888</a>]  Camarillo, G. and H. Schulzrinne, "The Session Description
              Protocol (SDP) Grouping Framework", <a href="./rfc5888">RFC 5888</a>,
              DOI 10.17487/RFC5888, June 2010,
              &lt;<a href="https://www.rfc-editor.org/info/rfc5888">https://www.rfc-editor.org/info/rfc5888</a>&gt;.



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   [<a id="ref-RFC5905">RFC5905</a>]  Mills, D., Martin, J., Ed., Burbank, J., and W. Kasch,
              "Network Time Protocol Version 4: Protocol and Algorithms
              Specification", <a href="./rfc5905">RFC 5905</a>, DOI 10.17487/RFC5905, June 2010,
              &lt;<a href="https://www.rfc-editor.org/info/rfc5905">https://www.rfc-editor.org/info/rfc5905</a>&gt;.

   [<a id="ref-RFC6051">RFC6051</a>]  Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
              Flows", <a href="./rfc6051">RFC 6051</a>, DOI 10.17487/RFC6051, November 2010,
              &lt;<a href="https://www.rfc-editor.org/info/rfc6051">https://www.rfc-editor.org/info/rfc6051</a>&gt;.

   [<a id="ref-RFC7201">RFC7201</a>]  Westerlund, M. and C. Perkins, "Options for Securing RTP
              Sessions", <a href="./rfc7201">RFC 7201</a>, DOI 10.17487/RFC7201, April 2014,
              &lt;<a href="https://www.rfc-editor.org/info/rfc7201">https://www.rfc-editor.org/info/rfc7201</a>&gt;.

   [<a id="ref-RFC7667">RFC7667</a>]  Westerlund, M. and S. Wenger, "RTP Topologies", <a href="./rfc7667">RFC 7667</a>,
              DOI 10.17487/RFC7667, November 2015,
              &lt;<a href="https://www.rfc-editor.org/info/rfc7667">https://www.rfc-editor.org/info/rfc7667</a>&gt;.

   [<a id="ref-RFC8174">RFC8174</a>]  Leiba, B., "Ambiguity of Uppercase vs Lowercase in
              <a href="./rfc2119">RFC 2119</a> Key Words", <a href="https://www.rfc-editor.org/bcp/bcp14">BCP 14</a>, <a href="./rfc8174">RFC 8174</a>,
              DOI 10.17487/RFC8174, May 2017,
              &lt;<a href="https://www.rfc-editor.org/info/rfc8174">https://www.rfc-editor.org/info/rfc8174</a>&gt;.

   [<a id="ref-RFC8285">RFC8285</a>]  Singer, D., Desineni, H., and R. Even, Ed., "A General
              Mechanism for RTP Header Extensions", <a href="./rfc8285">RFC 8285</a>,
              DOI 10.17487/RFC8285, October 2017,
              &lt;<a href="https://www.rfc-editor.org/info/rfc8285">https://www.rfc-editor.org/info/rfc8285</a>&gt;.

<span class="h3"><a class="selflink" id="section-9.2" href="#section-9.2">9.2</a>.  Informative References</span>

   [<a id="ref-RFC3711">RFC3711</a>]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              <a href="./rfc3711">RFC 3711</a>, DOI 10.17487/RFC3711, March 2004,
              &lt;<a href="https://www.rfc-editor.org/info/rfc3711">https://www.rfc-editor.org/info/rfc3711</a>&gt;.

   [<a id="ref-RFC5506">RFC5506</a>]  Johansson, I. and M. Westerlund, "Support for Reduced-Size
              Real-Time Transport Control Protocol (RTCP): Opportunities
              and Consequences", <a href="./rfc5506">RFC 5506</a>, DOI 10.17487/RFC5506,
              April 2009, &lt;<a href="https://www.rfc-editor.org/info/rfc5506">https://www.rfc-editor.org/info/rfc5506</a>&gt;.

   [<a id="ref-RFC6285">RFC6285</a>]  Ver Steeg, B., Begen, A., Van Caenegem, T., and Z. Vax,
              "Unicast-Based Rapid Acquisition of Multicast RTP
              Sessions", <a href="./rfc6285">RFC 6285</a>, DOI 10.17487/RFC6285, June 2011,
              &lt;<a href="https://www.rfc-editor.org/info/rfc6285">https://www.rfc-editor.org/info/rfc6285</a>&gt;.

   [<a id="ref-RFC6828">RFC6828</a>]  Xia, J., "Content Splicing for RTP Sessions", <a href="./rfc6828">RFC 6828</a>,
              DOI 10.17487/RFC6828, January 2013,
              &lt;<a href="https://www.rfc-editor.org/info/rfc6828">https://www.rfc-editor.org/info/rfc6828</a>&gt;.




<span class="grey">Xia, et al.                  Standards Track                   [Page 21]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-22" ></span>
<span class="grey"><a href="./rfc8286">RFC 8286</a>                RTP Splicing Notification           October 2017</span>


   [<a id="ref-RFC6904">RFC6904</a>]  Lennox, J., "Encryption of Header Extensions in the Secure
              Real-time Transport Protocol (SRTP)", <a href="./rfc6904">RFC 6904</a>,
              DOI 10.17487/RFC6904, April 2013,
              &lt;<a href="https://www.rfc-editor.org/info/rfc6904">https://www.rfc-editor.org/info/rfc6904</a>&gt;.

   [<a id="ref-RFC8126">RFC8126</a>]  Cotton, M., Leiba, B., and T. Narten, "Guidelines for
              Writing an IANA Considerations Section in RFCs", <a href="https://www.rfc-editor.org/bcp/bcp26">BCP 26</a>,
              <a href="./rfc8126">RFC 8126</a>, DOI 10.17487/RFC8126, June 2017,
              &lt;<a href="https://www.rfc-editor.org/info/rfc8126">https://www.rfc-editor.org/info/rfc8126</a>&gt;.

   [<a id="ref-SCTE35">SCTE35</a>]   Society of Cable Telecommunications Engineers (SCTE),
              "Digital Program Insertion Cueing Message for Cable",
              2016, &lt;<a href="http://www.scte.org/SCTEDocs/Standards/SCTE%2035%202016.pdf">http://www.scte.org/SCTEDocs/Standards/</a>
              <a href="http://www.scte.org/SCTEDocs/Standards/SCTE%2035%202016.pdf">SCTE%2035%202016.pdf</a>&gt;.

Acknowledgements

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

Authors' Addresses

   Jinwei Xia
   Huawei

   Email: xiajinwei@huawei.com


   Roni Even
   Huawei

   Email: roni.even@huawei.com


   Rachel Huang
   Huawei

   Email: rachel.huang@huawei.com


   Lingli Deng
   China Mobile

   Email: denglingli@chinamobile.com






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