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<rfc xmlns:xi="http://www.w3.org/2001/XInclude" ipr="trust200902" docName="draft-ietf-avtcore-rtp-scip-09" number="9607" category="std" consensus="true" submissionType="IETF"> submissionType="IETF" updates="" obsoletes="" tocInclude="true" xml:lang="en" version="3" sortRefs="true" symRefs="true">

  <front>
    <title abbrev="SCIP RTP Payload Format">RTP Payload Format for the Secure Communication
   Interoperability Protocol (SCIP) Codec</title>
    <seriesInfo name="RFC" value="9607"/>
    <author initials="D." surname="Hanson" fullname="Daniel Hanson">
      <organization>General Dynamics Mission Systems, Inc.</organization>
      <address>
        <postal>
          <street>150 Rustcraft Road</street>
          <city>Dedham</city>
          <region>MA</region>
          <code>02026</code>
          <country>United States of America</country>
        </postal>
        <email>dan.hanson@gd-ms.com</email>
      </address>
    </author>
    <author initials="M." surname="Faller" fullname="Michael Faller">
      <organization>General Dynamics Mission Systems, Inc.</organization>
      <address>
        <postal>
          <street>150 Rustcraft Road</street>
          <city>Dedham</city>
          <region>MA</region>
          <code>02026</code>
          <country>United States of America</country>
        </postal>
        <email>michael.faller@gd-ms.com</email>
	<email>MichaelFFaller@gmail.com</email>
      </address>
    </author>
    <author initials="K." surname="Maver" fullname="Keith Maver">
      <organization>General Dynamics Mission Systems, Inc.</organization>
      <address>
        <postal>
          <street>150 Rustcraft Road</street>
          <city>Dedham</city>
          <region>MA</region>
          <code>02026</code>
          <country>United States of America</country>
        </postal>
        <email>keith.maver@gd-ms.com</email>
      </address>
    </author>
    <date year="2024" month="February" day="13"/>

    <workgroup>Payload Working Group</workgroup> month="July"/>
    <area>WIT</area>
    <workgroup>avtcore</workgroup>

<keyword>SCIP</keyword>
<keyword>FNBDT</keyword>
<keyword>NATO</keyword>
<keyword>Department of Defense</keyword>
<keyword>DoD</keyword>
<keyword>NSA</keyword>
<keyword>STANAG</keyword>
<keyword>ICWG</keyword>
<keyword>IICWG</keyword>
<keyword>SCIPWG</keyword>

    <abstract>
      <t>This document describes the RTP payload format of the Secure
      Communication Interoperability Protocol (SCIP). SCIP is an application
 layer
      application-layer protocol that provides end-to-end capability exchange,
 packetization/de-packetization of media, reliable transport, and session
      establishment, payload encryption.</t>

<t>SCIP handles packetization/de-packetization encryption, packetization and de-packetization of the encrypted media
      media, and acts as reliable transport.  This document provides a
 tunneling protocol, treating SCIP payloads as opaque octets to globally
      available reference that can be
 encapsulated within RTP payloads prior to transmission or decapsulated
 on reception. SCIP payloads are sized to fit within used for the maximum transmission unit (MTU)
 when transported over RTP thereby avoiding fragmentation.</t>

<t>SCIP transmits encrypted traffic development of network
      equipment and does not require the use procurement of Secure RTP
 (SRTP) for payload protection.  SCIP also provides for reliable transport at
 the application layer, so it is not necessary to negotiate RTCP retransmission capabilities.</t>

<t>To establish reliable communications using SCIP over RTP, it is important services that middle boxes avoid parsing or modifying SCIP payloads.
 Because support SCIP payloads are confidentiality traffic. The
      intended audience is network security policymakers; network
      administrators, architects, and integrity protected original equipment manufacturers (OEMs);
      procurement personnel; and are only decipherable by
 the originating government agency and receiving SCIP devices, modification of the payload by
 middle boxes would be detected as an integrity failure in SCIP devices,
 resulting in retransmission and/or communication failure. Middle
 boxes do not need to parse the SCIP payloads to correctly transport them.
 Not only commercial industry
      representatives.</t>
    </abstract>
    <note>
      <name>IESG Note</name>
      <t>This IETF specification depends upon a second technical specification
      that is parsing unnecessary not available publicly, namely <xref target="SCIP210"/>.
	  The IETF was therefore unable to tunnel/detunnel SCIP within RTP,
 but the parsing and filtering conduct a security review of SCIP payloads by middle boxes would likely lead that
      specification, independently or when carried inside Audio/Video
      Transport (AVT). Implementers need to
 ossification be aware that the IETF hence
      cannot verify any of the evolving SCIP protocol.</t>

    </abstract> security claims contained in this document.</t>
    </note>
  </front>
  <middle>
    <section anchor="key-points"><name>Key anchor="key-points">
      <name>Key Points</name>
      <!-- section 1 -->
      <ul>
        <li>SCIP is an application layer application-layer protocol that uses RTP as a transport.  This document defines
 the SCIP media subtypes to be listed in the Session Description Protocol (SDP) and only requires
 a basic RTP transport channel for SCIP payloads. This basic transport channel is comparable to
 Clearmode as defined by <xref target="RFC4040"/> Clearmode.</li> target="RFC4040"/>.</li>
        <li>SCIP transmits encrypted traffic and does not require the use of Secure RTP
 (SRTP) for payload protection.  SCIP also provides for reliable transport at
 the application layer, so it is not necessary to negotiate RTCP retransmission capabilities.</li>
        <li>SCIP includes built-in mechanisms that negotiate protocol message versions and capabilities.
 To avoid SCIP protocol ossification (as described in <xref target="RFC9170"/>), it is important
 for middleboxes to not attempt parsing of the SCIP payload. As described in this document,
 such parsing serves no useful purpose.</li>
        <li>SCIP is designed to be network agnostic. It can operate over any digital link, including
 non-IP modem-based PSTN and ISDN. The SCIP media subtypes listed in this document were
 developed for SCIP to operate over RTP.</li>
        <li>SCIP handles packetization/de-packetization packetization and de-packetization of payloads by producing encrypted media packets
 that are not greater than the MTU size. The SCIP payload is opaque to the network, therefore, SCIP functions
 as a tunneling protocol for the encrypted media, without the need for middle boxes middleboxes to parse SCIP payloads.
 Since SCIP payloads are integrity protected, modification of the SCIP payload is detected as an
 integrity violation by SCIP endpoints endpoints, leading to communication failure.</li>

<li>SCIP includes built-in mechanisms that negotiate protocol message versions and capabilities.
 To avoid SCIP protocol ossification (as described in <xref target="RFC9170"/>), it is important
 for middle boxes to not attempt parsing of the SCIP payload. As described in this document,
 such parsing serves no useful purpose.</li></ul>
      </ul>
    </section>
    <section anchor="introduction"><name>Introduction</name> anchor="introduction">
      <name>Introduction</name>
      <!-- section 2 -->

<t>
The purpose of this document is to provide enough information to enable SCIP payloads to be transported
 through the network without modification or filtering. The document provides a reference for network
 security policymakers; network equipment OEMs, administrators, and architects; procurement personnel;
 and government agency and commercial industry representatives.
</t>

<t>
The

      <t>This document details usage of the "audio/scip" and "video/scip"
      pseudo-codecs <xref target="AUDIOSCIP"/>,
 <xref target="VIDEOSCIP"/> target="MediaTypes"/> as a secure session establishment protocol and media
      transport protocol over RTP.
 It RTP.</t>
	  <t>It discusses (1) how encrypted how:</t>
      <ol spacing="normal">
	<li>encrypted audio and video codec payloads are transported over RTP;
 (2) the RTP;</li>
	<li>the IP network layer does not implementing implement SCIP as a protocol since
    SCIP operates at the application layer in endpoints; (3) the endpoints;</li>
	<li>the IP network layer enabling enables SCIP traffic to transparently pass
       through the network; (4) network;</li>
	<li>some network devices do not recognizing SCIP, recognize SCIP and thus removing may remove the scip SCIP
       codecs from the SDP media payload declaration before forwarding
       to the next network node; and finally,
 (5) SCIP finally,</li>
	<li>SCIP endpoint devices do not operating operate on networks due to if the scip SCIP
       media subtype removal is removed from the SDP media payload declaration.
</t> declaration.</li>
      </ol>
      <t>The United States, along with its NATO Partners, have implemented SCIP in secure voice, video, and
 data products operating on commercial, private, and tactical IP networks
 worldwide using the scip media subtype. The SCIP data traversing the network is encrypted,
 and network equipment in-line with the session cannot interpret the traffic stream in any way.
 SCIP-based RTP traffic is opaque and can vary significantly in structure and frequency frequency, making
 traffic profiling not possible.  Also, as the SCIP protocol continues to evolve independently
 of this document, any network device that attempts to filter traffic (e.g., deep packet inspection)
 may cause unintended consequences in the future when changes to the SCIP traffic may not be recognized by
 the network device.
</t>
      <t>The SCIP protocol defined in SCIP-210 <xref target="SCIP210"/> includes built-in
 support for packetization/de-packetization, packetization and de-packetization, retransmission,
 capability exchange, version negotiation, and payload encryption. Since the traffic is encrypted,
 neither the RTP transport nor middle boxes middleboxes can usefully parse or modify SCIP
 payloads; modifications are detected as integrity violations resulting in
 retransmission, and eventually, communication failure.</t>
      <t>Because knowledge of the SCIP payload format is not needed to transport SCIP signaling or
 media through middle boxes, middleboxes, SCIP-210 represents an informative reference. While older versions
 of the SCIP-210 specification are publicly available, the authors strongly encourage
 network implementers to treat SCIP payloads as opaque octets. When handled correctly, such
 treatment does not require referring to SCIP-210, and any assumptions about the format of
 SCIP messages defined in SCIP-210 are likely to lead to protocol ossification and
 communication failures as the protocol evolves.</t>

      <aside>
        <t>Note: The IETF has not conducted a security review of SCIP and
        therefore has not verified the claims contained in this document.</t>
      </aside>

      <section anchor="conventions"><name>Conventions</name> anchor="conventions">
        <name>Conventions</name>
        <!-- section 2.1 -->

<t>The
        <t>
    The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL
   NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED",
   "MAY", "<bcp14>MUST</bcp14>", "<bcp14>MUST NOT</bcp14>",
    "<bcp14>REQUIRED</bcp14>", "<bcp14>SHALL</bcp14>", "<bcp14>SHALL NOT</bcp14>",
    "<bcp14>SHOULD</bcp14>", "<bcp14>SHOULD NOT</bcp14>",
    "<bcp14>RECOMMENDED</bcp14>", "<bcp14>NOT RECOMMENDED</bcp14>",
    "<bcp14>MAY</bcp14>", and "OPTIONAL" "<bcp14>OPTIONAL</bcp14>" in this document are to be
    interpreted as described in BCP 14 BCP&nbsp;14 <xref target="RFC2119"/> <xref
    target="RFC8174"/> when, and only when, they appear in all capitals, as
    shown here.</t>

<t>Best here.
        </t>
        <t>The best current practices for writing an RTP payload format
   specification were followed
   specification, as per <xref target="RFC2736"/> and <xref target="RFC8088"/>.</t> target="RFC8088"/>, were followed.</t>
        <t>When referring to the Secure Communication Interoperability
   Protocol, the uppercase acronym "SCIP" is used.  When referring
   to the media subtype scip, lowercase "scip" is used.</t>
      </section>
      <section anchor="abbreviations"><name>Abbreviations</name> anchor="abbreviations">
        <name>Abbreviations</name>
        <!-- section 2.2 -->
<t>The following abbreviations are used in this document.</t>
        <dl newline="false" indent="10" spacing="compact"> spacing="normal">
          <dt>AVP:</dt>     <dd>Audio/Video
          <dd>Audio-Visual Profile</dd>
          <dt>AVPF:</dt>    <dd>Audio/Video
          <dd>Audio-Visual Profile with Feedback</dd>
          <dt>FNBDT:</dt>
          <dd>Future Narrowband Digital Terminal</dd>
          <dt>ICWG:</dt>
          <dd>Interoperability Control Working Group</dd>
          <dt>IICWG:</dt>
          <dd>International Interoperability Control Working Group</dd>
          <dt>MELPe:</dt>
          <dd>Mixed Excitation Linear Prediction Enhanced</dd>
          <dt>MTU:</dt>
          <dd>Maximum Transmission Unit</dd>
          <dt>NATO:</dt>
          <dd>North Atlantic Treaty Organization</dd>
          <dt>OEM:</dt>
          <dd>Original Equipment Manufacturer</dd>
          <dt>SAVP:</dt>
          <dd>Secure Audio/Video Audio-Visual Profile</dd>
          <dt>SAVPF:</dt>
          <dd>Secure Audio/Video Audio-Visual Profile with Feedback</dd>
          <dt>SCIP:</dt>
          <dd>Secure Communication Interoperability Protocol</dd>
          <dt>SDP:</dt>
          <dd>Session Description Protocol</dd>
          <dt>SRTP:</dt>
          <dd>Secure Real-Time Real-time Transport Protocol</dd>
          <dt>STANAG:</dt>
          <dd>Standardization Agreement</dd>
        </dl>
      </section>
    </section>
    <section anchor="background"><name>Background</name> anchor="background">
      <name>Background</name>
      <!-- section 3 -->

<t>The Secure Communication Interoperability Protocol (SCIP)
   allows the negotiation of several voice, data, and video
   applications using various cryptographic suites. SCIP also provides several
   important characteristics that have led to its broad acceptance as a secure
   communications protocol.</t>
      <t>SCIP began in the United States as the Future Narrowband Digital
   Terminal (FNBDT) Protocol in the late 1990s.  A combined U.S. Department of Defense
   and vendor consortium formed a governing organization named the
   Interoperability Control Working Group (ICWG) to manage the
   protocol.  In time, the group expanded to include NATO, NATO
   partners
   partners, and European vendors under the name International
   Interoperability Control Working Group (IICWG), which was later
   renamed the SCIP Working Group.</t>
      <t>First generation SCIP devices operated on circuit-switched
   networks.  SCIP was then expanded to radio and IP networks.
   The scip media subtype transports SCIP secure session
   establishment signaling and secure application traffic.  The
   built-in negotiation and flexibility provided by the SCIP
   protocols make it a natural choice for many scenarios that
   require various secure applications and associated encryption
   suites.  SCIP has been adopted by NATO in STANAG 5068.
   SCIP standards are currently available to participating
   government/military
   government and military communities and select OEMs of equipment
   that support SCIP.</t>
      <t>However, SCIP must operate over global networks (including
   private and commercial networks).  Without access to necessary
   information to support SCIP, some networks may not support the
   SCIP media subtypes.  Issues may occur simply because
   information is not as readily available to OEMs, network
   administrators, and network architects.</t>
      <t>This document provides essential information about audio/scip the "audio/scip" and
   video/scip
   "video/scip" media subtypes that enables enable network equipment
   manufacturers to include settings for "scip" as a known audio and video media
   subtype in their equipment. This enables network administrators
   to define and implement a compatible security policy which that includes audio and
   video media subtypes "audio/scip" and "video/scip", respectively, as permitted
   codecs on the network.</t>
      <t>All current IP-based SCIP endpoints implement "scip" as a media
   subtype.  Registration of scip as a media subtype provides a
   common reference for network equipment manufacturers to
   recognize SCIP in an SDP payload declaration.</t>
    </section>
    <section anchor="media-format-description"><name>Payload anchor="media-format-description">
      <name>Payload Format</name>
      <!-- section 4 -->

      <t>The "scip" media subtype identifies and indicates support for and identifies SCIP
      traffic that is being transported over RTP.  Transcoding, lossy
      compression, or other data modifications MUST NOT <bcp14>MUST NOT</bcp14> be
      performed by the network on the SCIP RTP payload.  The audio/scip "audio/scip" and
   video/scip
      "video/scip" media subtype data streams within the network, including the
      VoIP network, MUST <bcp14>MUST</bcp14> be a transparent relay and be treated
      as "clear-channel data", similar to the Clearmode media subtype defined
      by <xref target="RFC4040"/>.</t>

<t>RFC 4040
      <t><xref target="RFC4040"/> is referenced because Clearmode does not define
   specific RTP payload content, packet size, or packet intervals, but rather
   enables Clearmode devices to signal that they support a compatible mode of
   operation and defines a transparent channel on which devices may communicate.
   This document takes a similar approach. Network devices that implement support for
   SCIP need to enable SCIP endpoints to signal that they support SCIP and
   provide a transparent channel on which SCIP endpoints may communicate.
</t>
      <t>SCIP is an application layer application-layer protocol that is defined in SCIP-210.
The SCIP traffic consists of encrypted SCIP control messages
and codec data. The payload size and interval will vary considerably depending on
the state of the SCIP protocol within the SCIP device.</t>

<t>Figure 1
      <t><xref target="fig-1"/> below illustrates the RTP payload format for SCIP.</t>
      <figure anchor="fig-1" align="left" suppress-title="false" pn="figure-1"> anchor="fig-1">
        <name slugifiedName="scip-payload">SCIP RTP Payload Format</name>
<artwork>
        <artwork align="left"><![CDATA[
 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                           RTP Header                          |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
|                                                               |
|                          SCIP payload Payload                         |
|                                                               |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
</artwork>
]]></artwork>
      </figure>
      <t>The SCIP codec produces an encrypted bitstream that is transported over RTP. Unlike other
codecs, SCIP does not have its own upper layer syntax (e.g., no Network Adaptation Layer (NAL)
units), but rather encrypts the output of the audio/video audio and video codecs that it uses
(e.g., G.729D, H.264 <xref target="RFC6184"/>, etc.).
SCIP achieves this by encapsulating the encrypted codec output that has been previously formatted
according to the relevant RTP payload specification for that codec. SCIP endpoints MAY <bcp14>MAY</bcp14> employ
mechanisms, such as Inter-media inter-media RTP Synchronization synchronization as described in <xref target="RFC8088"/> Section 3.3.4, target="RFC8088" sectionFormat="comma" section="3.3.4"/>, to
synchronize audio/scip "audio/scip" and video/scip "video/scip" streams.</t>

<t>Figure 2
      <t><xref target="fig-2"/> below illustrates notionally how codec packets and SCIP control
messages are packetized for transmission over RTP.</t>
      <figure anchor="fig-2" align="left" suppress-title="false" pn="figure-2"> anchor="fig-2">
        <name slugifiedName="scip-architecture">SCIP RTP Architecture</name>
<artwork>
        <artwork align="left"><![CDATA[
+-----------+              +-----------------------+
|   Codec   |              | SCIP control messages |
+-----------+              +-----------------------+
      |                                |
      |                                |
      V                                V
+--------------------------------------------------+
|             Packetizer* (&lt;= (<= MTU size)            |
+--------------------------------------------------+
          |                        |
          |                        |
          V                        |
  +--------------+                 |
  |  Encryption  |                 |
  +--------------+                 |
          |                        |
          |                        |
          V                        V
+--------------------------------------------------+
|                      RTP                         |
+--------------------------------------------------+
</artwork>
]]></artwork>
      </figure>

<aside><t>* Packetizer: The
      <dl>
        <dt>* Packetizer:</dt><dd>The SCIP application layer will ensure that all traffic sent to
 the RTP layer will not exceed the MTU size.  The receiving SCIP RTP layer will handle
 packet identification, ordering, and reassembly.   When required, the SCIP application
 layer handles error detection and retransmission.
</t></aside> retransmission.</dd>
      </dl>
      <t>As described above, the SCIP RTP payload format is variable and cannot be described in
specificity in this document. Details can be found in SCIP-210.
SCIP will continue to evolve and and, as such such, the SCIP RTP traffic MUST NOT <bcp14>MUST NOT</bcp14> be filtered
by network devices based upon what currently is observed or documented. The focus of this document is for
network devices to consider the SCIP RTP payload as opaque and allow it to traverse the
network. Network devices MUST NOT <bcp14>MUST NOT</bcp14> modify SCIP RTP packets.</t>
      <section anchor="rtp-header-fields"><name>RTP anchor="rtp-header-fields">
        <name>RTP Header Fields</name>
        <!-- section 4.1 -->

<t>The SCIP RTP header fields SHALL <bcp14>SHALL</bcp14> conform to RFC 3550.</t> <xref target="RFC3550"/>.</t>
        <t>SCIP traffic may be continuous or discontinuous.  The Timestamp
   field MUST <bcp14>MUST</bcp14> increment based on the sampling clock for
   discontinuous transmission as described in <xref target="RFC3550"/>, Section
   5.1. target="RFC3550" sectionFormat="comma" section="5.1"/>.  The Timestamp field for continuous transmission
   applications is dependent on the sampling rate of the media as
   specified in the media subtype's specification (e.g., MELPe). Mixed Excitation Linear Prediction Enhanced (MELPe)).
   Note that during a SCIP session, both discontinuous and
   continuous traffic are highly probable.</t>
        <t>The Marker bit SHALL <bcp14>SHALL</bcp14> be set to zero for discontinuous traffic.
   The Marker bit for continuous traffic is based on the
   underlying media subtype specification.  The underlying media
   is opaque within SCIP RTP packets.</t>
      </section>
      <section anchor="congestion-control"><name>Congestion anchor="congestion-control">
        <name>Congestion Control Considerations</name>
        <!-- section 4.2 -->

<t>The bitrate of SCIP may be adjusted depending on the capability of the underlying
   codec (such as MELPe <xref target="RFC8130"/>, G.729D <xref target="RFC3551"/>, etc.).
   The number of encoded audio frames per packet may
   also be adjusted to control congestion.  Discontinuous transmission may also
   be used if supported by the underlying codec.
</t>
        <t>
Since UDP does not provide congestion control, applications that use
RTP over UDP SHOULD <bcp14>SHOULD</bcp14> implement their own congestion control above the
UDP layer <xref target="RFC8085"/> and MAY <bcp14>MAY</bcp14> also implement a transport circuit
breaker <xref target="RFC8083"/>. Work in the RTP Media Congestion Avoidance Techniques
(RMCAT) working group <xref target="RMCAT"/> describes
the interactions and conceptual interfaces necessary between the
application components that relate to congestion control, including
the RTP layer, the higher-level media codec control layer, and the
lower-level transport interface, as well as components dedicated to
congestion control functions.
</t>
        <t>Use of the packet loss feedback mechanisms in AVPF <xref target="RFC4585"/> and
 SAVPF <xref target="RFC5124"/> are OPTIONAL <bcp14>OPTIONAL</bcp14> because SCIP itself manages retransmissions
 of some errored or lost packets. Specifically, the Payload-Specific Feedback Messages payload-specific feedback messages
 defined in RFC 4585 section 6.3 <xref target="RFC4585" sectionFormat="comma" section="6.3"/> are OPTIONAL <bcp14>OPTIONAL</bcp14> when transporting video data.
</t>
      </section>
      <section anchor="augmented-protocols"><name>Use anchor="augmented-protocols">
        <name>Use of Augmented RTP Transport Protocols  RTPs with SCIP</name>
        <!-- section 4.3 -->

<t>The SCIP application layer application-layer protocol uses RTP as a basic transport for the audio/scip "audio/scip" and
 video/scip
 "video/scip" payloads. Additional RTP transport protocols RTPs that do not modify the SCIP payload
 are considered OPTIONAL <bcp14>OPTIONAL</bcp14> in this document and are discretionary for a SCIP device vendor to implement.
 Some examples include include, but are not limited to:</t>
        <ul>
<li>RTP
          <li>"RTP Payload Format for Generic Forward Error Correction Correction" <xref target="RFC5109"/></li>
<li>Multiplexing
          <li>"Multiplexing RTP Data and Control Packets on a Single Port Port" <xref target="RFC5761"/></li>
<li>Symmetric RTP/RTP
          <li>"Symmetric RTP / RTP Control Protocol (RTCP) (RTCP)" <xref target="RFC4961"/></li>
<li>Negotiating
          <li>"Negotiating Media Multiplexing Using the Session Description Protocol (BUNDLE) (SDP)" a.k.a.&nbsp;BUNDLE <xref target="RFC9143"/></li>
        </ul>
      </section>
    </section>
    <section anchor="payload-format-parameters"><name>Payload anchor="payload-format-parameters">
      <name>Payload Format Parameters</name>
      <!-- section 5 -->

<t>The SCIP RTP payload format is identified using the scip media
   subtype, which is registered in accordance with <xref target="RFC4855"/> and
   per the media type registration template form from <xref target="RFC6838"/>.  A
   clock rate of 8000 Hz SHALL <bcp14>SHALL</bcp14> be used for "audio/scip".  A clock
   rate of 90000 Hz SHALL <bcp14>SHALL</bcp14> be used for "video/scip".</t>
      <section anchor="media-subtype-audioscip"><name>Media anchor="media-subtype-audioscip">

        <name>Media Subtype "audio/scip"</name>
        <!-- section 5.1 -->

<t>Media type name: audio</t>

<t>Media subtype name: scip</t>

<t>Required parameters: N/A</t>

<t>Optional parameters: N/A</t>

<t>Encoding considerations: Binary.
	<dl>
	  <dt>Type name:</dt><dd>audio</dd>
	  <dt>Subtype name:</dt><dd>scip</dd>
	  <dt>Required parameters:</dt><dd>N/A</dd>
	  <dt>Optional parameters:</dt><dd>N/A</dd>
	  <dt>Encoding considerations:</dt><dd>Binary.  This media subtype is
	  only defined for transfer via RTP.  There SHALL <bcp14>SHALL</bcp14> be no
   encoding/decoding (transcoding)
	  transcoding of the audio stream as it traverses
	  the network.</t>

<t>Security considerations: See Section 7.</t>

<t>Interoperability considerations: N/A</t>

<t>Published specifications: <xref target="SCIP210"/></t>

<t>Applications which network.</dd>
	  <dt>Security considerations:</dt><dd>See <xref target="security-considerations"/>.</dd>
	  <dt>Interoperability considerations:</dt><dd>N/A</dd>
	  <dt>Published specification:</dt><dd><xref target="SCIP210"/></dd>
	  <dt>Applications that use this media: N/A</t>

<t>Fragment Identifier considerations: none</t>

<t>Restrictions on usage: N/A</t>

<t>Additional information:</t>

<t indent="3">1. Deprecated media type:</dt><dd>N/A</dd>
	  <dt>Fragment identifier considerations:</dt><dd>none</dd>
	  <dt>Additional information:</dt><dd>
	  <t><br/></t>
	  <dl spacing="compact">
	    <dt>Deprecated alias names for this type: N/A</t>

<t indent="3">2. Magic number(s): N/A</t>

<t indent="3">3. File extension(s): N/A</t>

<t indent="3">4. Macintosh type:</dt><dd>N/A</dd>
	    <dt>Magic number(s):</dt><dd>N/A</dd>
	    <dt>File extension(s):</dt><dd>N/A</dd>
	    <dt>Macintosh file type code: N/A</t>

<t indent="3">5. Object Identifiers: N/A</t>

<t>Person code(s):</dt><dd>N/A</dd>
	  </dl>
	</dd>
	<dt>Person &amp; email address to contact for further information:</t>

<t indent="3">1. Name: Michael
	information:</dt><dd>Michael Faller (michael.faller@gd-ms.com or MichaelFFaller@gmail.com) and
	Daniel Hanson</t>

<t indent="3">2. Email: michael.faller@gd-ms.com and dan.hanson@gd-ms.com</t>

<t>Intended usage: Common</t>

<t>Authors:</t>

<t indent="3">Michael Hanson (dan.hanson@gd-ms.com)</dd>
	<dt>Intended usage:</dt><dd>COMMON</dd>
	<dt>Restrictions on usage:</dt><dd>N/A</dd>
	<dt>Authors:</dt><dd>Michael Faller - michael.faller@gd-ms.com</t>

<t indent="3">Daniel (michael.faller@gd-ms.com or MichaelFFaller@gmail.com) and Daniel Hanson - dan.hanson@gd-ms.com</t>

<t>Change controller:</t>

<t indent="3">SCIP (dan.hanson@gd-ms.com)</dd>
	<dt>Change controller:</dt><dd>SCIP Working Group - ncia.cis3@ncia.nato.int</t> (ncia.cis3@ncia.nato.int)</dd>
	</dl>
      </section>

      <section anchor="media-subtype-videoscip"><name>Media anchor="media-subtype-videoscip">
      <name>Media Subtype "video/scip"</name>
<!-- section 5.2 -->

<t>Media type name: video</t>

<t>Media subtype name: scip</t>

<t>Required parameters: N/A</t>

<t>Optional parameters: N/A</t>

<t>Encoding considerations: Binary.
      <dl>
	<dt>Type name:</dt><dd>video</dd>
	<dt>Subtype name:</dt><dd>scip</dd>
	<dt>Required parameters:</dt><dd>N/A</dd>
	<dt>Optional parameters:</dt><dd>N/A</dd>
	<dt>Encoding considerations:</dt><dd>Binary.  This media subtype is
	only defined for transfer via RTP.  There SHALL <bcp14>SHALL</bcp14> be no
   encoding/decoding (transcoding)
	transcoding of the video stream as it traverses
	the network.</t>

<t>Security considerations: See Section 7.</t>

<t>Interoperability considerations: N/A</t>

<t>Published specifications: <xref target="SCIP210"/></t>

<t>Applications which network.</dd>
	<dt>Security considerations:</dt><dd>See <xref target="security-considerations"/>.</dd>
	<dt>Interoperability considerations:</dt><dd>N/A</dd>
	<dt>Published specification:</dt><dd><xref target="SCIP210"/></dd>
	<dt>Applications that use this media: N/A</t>

<t>Fragment Identifier considerations: none</t>

<t>Restrictions on usage: N/A</t>

<t>Additional information:</t>

<t indent="3">1. Deprecated media type:</dt><dd>N/A</dd>
	<dt>Fragment identifier considerations:</dt><dd>none</dd>
	<dt>Additional information:</dt><dd>
	<t><br/></t>
	<dl spacing="compact">
	  <dt>Deprecated alias names for this type: N/A</t>

<t indent="3">2. Magic number(s): N/A</t>

<t indent="3">3. File extension(s): N/A</t>

<t indent="3">4. Macintosh type:</dt><dd>N/A</dd>
	  <dt>Magic number(s):</dt><dd>N/A</dd>
	  <dt>File extension(s):</dt><dd>N/A</dd>
	  <dt>Macintosh file type code: N/A</t>

<t indent="3">5. Object Identifiers: N/A</t>

<t>Person code(s):</dt><dd>N/A</dd>
	</dl>
      </dd>
      <dt>Person &amp; email address to contact for further information:</t>

<t indent="3">1. Name: Michael information:</dt><dd>Michael
      Faller (michael.faller@gd-ms.com or MichaelFFaller@gmail.com) and Daniel Hanson</t>

<t indent="3">2. Email: michael.faller@gd-ms.com and dan.hanson@gd-ms.com</t>

<t>Intended usage: Common</t>

<t>Authors:</t>

<t indent="3">Michael Hanson
      (dan.hanson@gd-ms.com)</dd>
      <dt>Intended usage:</dt><dd>COMMON</dd>
      <dt>Restrictions on usage:</dt><dd>N/A</dd>
      <dt>Authors:</dt><dd>Michael Faller - michael.faller@gd-ms.com</t>

<t indent="3">Daniel (michael.faller@gd-ms.com or MichaelFFaller@gmail.com) and Daniel Hanson - dan.hanson@gd-ms.com</t>

<t>Change controller:</t>

<t indent="3">SCIP (dan.hanson@gd-ms.com)</dd>
      <dt>Change controller:</dt><dd>SCIP Working Group - ncia.cis3@ncia.nato.int</t> (ncia.cis3@ncia.nato.int)</dd>
      </dl>
    </section>

      <section anchor="mapping-to-sdp"><name>Mapping anchor="mapping-to-sdp">
        <name>Mapping to SDP</name>
        <!-- section 5.3 -->

<t>The mapping of the above defined above-defined payload format media subtype
   and its parameters SHALL <bcp14>SHALL</bcp14> be implemented according to Section 3
   of <xref target="RFC4855"/>.</t> target="RFC4855" sectionFormat="of" section="3"/>.</t>
        <t>Since SCIP includes its own facilities for capabilities exchange,
it is only necessary to negotiate the use of SCIP within SDP Offer/Answer;
the specific codecs to be encapsulated within SCIP are then negotiated via
the exchange of SCIP control messages.</t>
        <t>The information carried in the media type specification has a
specific mapping to fields in the Session Description Protocol (SDP)
<xref target="RFC8866"/>, which is commonly used to describe RTP sessions.
When SDP is used to specify sessions employing the SCIP codec, the mapping
is as follows:</t>
        <ul>
          <li>The media type ("audio") goes in SDP "m=" as the media name for audio/scip, "audio/scip",
and the media type ("video") goes in SDP "m=" as the media name for video/scip.</li> "video/scip".</li>
          <li>The media subtype ("scip") goes in SDP "a=rtpmap" as the encoding name.
The required parameter "rate" also goes in "a=rtpmap" as the clock rate.</li>
          <li>The optional parameters "ptime" and "maxptime" go in the SDP "a=ptime" and
"a=maxptime" attributes, respectively.</li>
        </ul>
        <t>An example mapping for audio/scip "audio/scip" is:</t>

<figure>
<artwork>
<![CDATA[
        <sourcecode type="sdp">
  m=audio 50000 RTP/AVP 96
  a=rtpmap:96 scip/8000]]>
  </artwork>
</figure> scip/8000
  </sourcecode>

        <t>An example mapping for video/scip "video/scip" is:</t>

<figure>
<artwork>
<![CDATA[
        <sourcecode type="sdp">
  m=video 50002 RTP/AVP 97
  a=rtpmap:97 scip/90000]]>
</artwork>
</figure> scip/90000
</sourcecode>

        <t>An example mapping for both audio/scip "audio/scip" and video/scip "video/scip" is:</t>

<figure>
<artwork>
<![CDATA[
        <sourcecode type="sdp">
  m=audio 50000 RTP/AVP 96
  a=rtpmap:96 scip/8000
  m=video 50002 RTP/AVP 97
  a=rtpmap:97 scip/90000]]>
</artwork>
</figure> scip/90000
</sourcecode>

      </section>
      <section anchor="sdp-offeranswer-considerations"><name>SDP anchor="sdp-offeranswer-considerations">
        <name>SDP Offer/Answer Considerations</name>
        <!-- section 5.4 -->

<t>In accordance with the SDP Offer/Answer model <xref target="RFC3264"/>, the
   SCIP device SHALL <bcp14>SHALL</bcp14> list the SCIP payload type number in order of
   preference in the "m" media line.</t>
        <t>For example, an SDP Offer with scip as the preferred audio media subtype:</t>

<figure>
<artwork>
<![CDATA[
        <sourcecode type="sdp">
  m=audio 50000 RTP/AVP 96 0 8
  a=rtpmap:96 scip/8000
  a=rtpmap:0 PCMU/8000
  a=rtpmap:8 PCMA/8000]]>
  </artwork>
</figure> PCMA/8000
</sourcecode>

      </section>
    </section>
    <section anchor="security-considerations"><name>Security anchor="security-considerations">
      <name>Security Considerations</name>
      <!-- section 6 -->

<t>RTP packets using the payload format defined in this
   specification are subject to the security considerations
   discussed in the RTP specification <xref target="RFC3550"/>, and in any
   applicable RTP profile such as RTP/AVP <xref target="RFC3551"/>, RTP/AVPF
   <xref target="RFC4585"/>, RTP/SAVP <xref target="RFC3711"/>, or RTP/SAVPF <xref target="RFC5124"/>.
   However, as "Securing the RTP Protocol Framework: Why RTP Does
   Not Mandate a Single Media Security Solution" <xref target="RFC7202"/>
   discusses, it is not an RTP payload format's responsibility to
   discuss or mandate what solutions are used to meet the basic
   security goals like confidentiality, integrity, and source
   authenticity for RTP in general.  This responsibility lies on
   anyone using RTP in an application.  They can find guidance on
   available security mechanisms and important considerations in
   "Options for Securing RTP Sessions" <xref target="RFC7201"/>.
   Applications SHOULD <bcp14>SHOULD</bcp14> use one or more appropriate strong security mechanisms.
   The rest of this Security Considerations section discusses the
   security impacting properties of the payload format itself.</t>
      <t>This RTP payload format and its media decoder do not exhibit
   any significant non-uniformity in the receiver-side
   computational complexity for packet processing, and thus do not
   inherently pose a denial-of-service threat due to the receipt
   of pathological data.  Nor data, nor does the RTP payload format contain
   any active content.</t>
      <t>SCIP only encrypts the contents transported in the RTP payload; it does not protect
  the RTP header or RTCP packets. Applications requiring additional RTP header headers and/or
  RTCP security might consider mechanisms such as SRTP <xref target="RFC3711"/>,
  however these additional mechanisms are considered OPTIONAL <bcp14>OPTIONAL</bcp14> in this document.</t>
    </section>
    <section anchor="iana-considerations"><name>IANA anchor="iana-considerations">
      <name>IANA Considerations</name>
      <!-- section 7 -->

<t>The audio/scip "audio/scip" and video/scip "video/scip" media subtypes have previously been
  registered with IANA <xref target="AUDIOSCIP"/> in the "Media Types" registry <xref target="VIDEOSCIP"/>. target="MediaTypes"/>. IANA should
   update <xref target="AUDIOSCIP"/> and <xref target="VIDEOSCIP"/> has updated
  these registrations to reference this document
   upon publication.</t> document.</t>
    </section>

    <section anchor="scip-contact-info"><name>SCIP anchor="scip-contact-info">
      <name>SCIP Contact Information</name>
      <!-- section 8 -->

<t>The SCIP protocol is maintained by the SCIP Working Group.  The current SCIP-210
specification <xref target="SCIP210"/> may be requested from the email address below.
</t>

<t>
   SCIP
    <contact>
      <organization>SCIP Working Group, CIS3 Partnership<br/>
   NATO Partnership</organization>
      <address>
        <postal>
	  <postalLine>NATO Communications and Information Agency<br/>
   Oude Agency</postalLine>
          <postalLine>Oude Waalsdorperweg 61<br/>
   2597 61</postalLine>
          <postalLine>2597 AK The Hague, Netherlands<br/>
   Email: ncia.cis3@ncia.nato.int</t> Netherlands</postalLine>
        </postal>
        <email>ncia.cis3@ncia.nato.int</email>
      </address>
    </contact>
      <t>An older public version of the SCIP-210 specification can be downloaded
 from <eref target="https://www.iad.gov/SecurePhone/index.cfm"/>.
</t>

</section>

  </middle>

  <back>

    <references title='Normative References'>

<reference anchor='RFC2119' target='https://www.rfc-editor.org/info/rfc2119'>
<front>
<title>Key words for use in RFCs to Indicate Requirement Levels</title>
<author fullname='S. Bradner' initials='S.' surname='Bradner'><organization/></author>
<date month='March' year='1997'/>
<abstract><t>In many standards track documents several words are used to signify the requirements in the specification.  These words are often capitalized. This document defines these words as they should be interpreted in IETF documents.  This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.</t></abstract>
</front>
<seriesInfo name='BCP' value='14'/>
<seriesInfo name='RFC' value='2119'/>
<seriesInfo name='DOI' value='10.17487/RFC2119'/>
</reference>

<reference anchor='RFC2736' target='https://www.rfc-editor.org/info/rfc2736'>
<front>
<title>Guidelines for Writers of RTP Payload Format Specifications</title>
<author fullname='M. Handley' initials='M.' surname='Handley'><organization/></author>
<author fullname='C. Perkins' initials='C.' surname='Perkins'><organization/></author>
<date month='December' year='1999'/>
<abstract><t>This document provides general guidelines aimed at assisting the authors of RTP Payload Format specifications in deciding on good formats.  This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.</t></abstract>
</front>
<seriesInfo name='BCP' value='36'/>
<seriesInfo name='RFC' value='2736'/>
<seriesInfo name='DOI' value='10.17487/RFC2736'/>
</reference>

<reference anchor='RFC3264' target='https://www.rfc-editor.org/info/rfc3264'>
<front>
<title>An Offer/Answer Model with Session Description Protocol (SDP)</title>
<author fullname='J. Rosenberg' initials='J.' surname='Rosenberg'><organization/></author>
<author fullname='H. Schulzrinne' initials='H.' surname='Schulzrinne'><organization/></author>
<date month='June' year='2002'/>
<abstract><t>This document defines a mechanism by which two entities can make use of the Session Description Protocol (SDP) to arrive at a common view of a multimedia session between them.  In the model, one participant offers the other a description of the desired session from their perspective, and the other participant answers with the desired session from their perspective.  This offer/answer model is most useful in unicast sessions where information from both participants is needed for the complete view of the session.  The offer/answer model is used by protocols like the Session Initiation Protocol (SIP).  [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name='RFC' value='3264'/>
<seriesInfo name='DOI' value='10.17487/RFC3264'/>
</reference>

<reference anchor='RFC3550' target='https://www.rfc-editor.org/info/rfc3550'>
<front>
<title>RTP: A Transport Protocol for Real-Time Applications</title>
<author fullname='H. Schulzrinne' initials='H.' surname='Schulzrinne'><organization/></author>
<author fullname='S. Casner' initials='S.' surname='Casner'><organization/></author>
<author fullname='R. Frederick' initials='R.' surname='Frederick'><organization/></author>
<author fullname='V. Jacobson' initials='V.' surname='Jacobson'><organization/></author>
<date month='July' year='2003'/>
<abstract><t>This memorandum describes RTP, the real-time transport protocol.  RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services.  RTP does not address resource reservation and does not guarantee quality-of- service for real-time services.  The data transport is augmented by a control protocol (RTCP) to allow monitoring of the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality.  RTP and RTCP are designed to be independent of the underlying transport and network layers.  The protocol supports the use of RTP-level translators and mixers. Most of the text in this memorandum is identical to RFC 1889 which it obsoletes.  There are no changes in the packet formats on the wire, only changes to the rules and algorithms governing how the protocol is used. The biggest change is an enhancement to the scalable timer algorithm for calculating when to send RTCP packets in order to minimize transmission in excess of the intended rate when many participants join a session simultaneously.  [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name='STD' value='64'/>
<seriesInfo name='RFC' value='3550'/>
<seriesInfo name='DOI' value='10.17487/RFC3550'/>
</reference>

<reference anchor='RFC3551' target='https://www.rfc-editor.org/info/rfc3551'>
<front>
<title>RTP Profile for Audio and Video Conferences with Minimal Control</title>
<author fullname='H. Schulzrinne' initials='H.' surname='Schulzrinne'><organization/></author>
<author fullname='S. Casner' initials='S.' surname='Casner'><organization/></author>
<date month='July' year='2003'/>
<abstract><t>This document describes a profile called &quot;RTP/AVP&quot; for the use of the real-time transport protocol (RTP), version 2, and the associated control protocol, RTCP, within audio and video multiparticipant conferences with minimal control.  It provides interpretations of generic fields within the RTP specification suitable for audio and video conferences.  In particular, this document defines a set U.S. Department of default mappings from payload type numbers to encodings. This document also describes how audio and video data may Defense Root Certificate should be carried within RTP.  It defines a set of standard encodings and their names when used within RTP.  The descriptions provide pointers
 installed to reference implementations and the detailed standards.  This document is meant as an aid for implementors of audio, video and other real-time multimedia applications. This memorandum obsoletes RFC 1890.  It is mostly backwards-compatible except for functions removed because two interoperable implementations were not found.  The additions to RFC 1890 codify existing practice in the use of payload formats under access this profile and include new payload formats defined since RFC 1890 was published.  [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name='STD' value='65'/>
<seriesInfo name='RFC' value='3551'/>
<seriesInfo name='DOI' value='10.17487/RFC3551'/>
</reference>

<reference anchor='RFC3711' target='https://www.rfc-editor.org/info/rfc3711'>
<front>
<title>The Secure Real-time Transport Protocol (SRTP)</title>
<author fullname='M. Baugher' initials='M.' surname='Baugher'><organization/></author>
<author fullname='D. McGrew' initials='D.' surname='McGrew'><organization/></author>
<author fullname='M. Naslund' initials='M.' surname='Naslund'><organization/></author>
<author fullname='E. Carrara' initials='E.' surname='Carrara'><organization/></author>
<author fullname='K. Norrman' initials='K.' surname='Norrman'><organization/></author>
<date month='March' year='2004'/>
<abstract><t>This document describes the Secure Real-time Transport Protocol (SRTP), a profile of the Real-time Transport Protocol (RTP), which can provide confidentiality, message authentication, and replay protection to the RTP traffic and to the control traffic for RTP, the Real-time Transport Control Protocol (RTCP).   [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name='RFC' value='3711'/>
<seriesInfo name='DOI' value='10.17487/RFC3711'/>
</reference>

<reference anchor='RFC4585' target='https://www.rfc-editor.org/info/rfc4585'>
<front>
<title>Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)</title>
<author fullname='J. Ott' initials='J.' surname='Ott'><organization/></author>
<author fullname='S. Wenger' initials='S.' surname='Wenger'><organization/></author>
<author fullname='N. Sato' initials='N.' surname='Sato'><organization/></author>
<author fullname='C. Burmeister' initials='C.' surname='Burmeister'><organization/></author>
<author fullname='J. Rey' initials='J.' surname='Rey'><organization/></author>
<date month='July' year='2006'/>
<abstract><t>Real-time media streams that use RTP are, to some degree, resilient against packet losses.  Receivers may use the base mechanisms of the Real-time Transport Control Protocol (RTCP) to report packet reception statistics and thus allow a sender to adapt its transmission behavior in the mid-term.  This is the sole means for feedback and feedback-based error repair (besides a few codec-specific mechanisms).  This document defines an extension to the Audio-visual Profile (AVP) that enables receivers to provide, statistically, more immediate feedback to the senders and thus allows for short-term adaptation and efficient feedback-based repair mechanisms to be implemented.  This early feedback profile (AVPF) maintains the AVP bandwidth constraints for RTCP and preserves scalability to large groups.  [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name='RFC' value='4585'/>
<seriesInfo name='DOI' value='10.17487/RFC4585'/>
</reference>

<reference anchor='RFC5124' target='https://www.rfc-editor.org/info/rfc5124'>
<front>
<title>Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)</title>
<author fullname='J. Ott' initials='J.' surname='Ott'><organization/></author>
<author fullname='E. Carrara' initials='E.' surname='Carrara'><organization/></author>
<date month='February' year='2008'/>
<abstract><t>An RTP profile (SAVP) for secure real-time communications and another profile (AVPF) to provide timely feedback from the receivers to a sender are defined in RFC 3711 and RFC 4585, respectively.  This memo specifies the combination of both profiles to enable secure RTP communications with feedback.  [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name='RFC' value='5124'/>
<seriesInfo name='DOI' value='10.17487/RFC5124'/>
</reference>

<reference anchor='RFC8174' target='https://www.rfc-editor.org/info/rfc8174'>
<front>
<title>Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words</title>
<author fullname='B. Leiba' initials='B.' surname='Leiba'><organization/></author>
<date month='May' year='2017'/>
<abstract><t>RFC 2119 specifies common key words that may be used in protocol  specifications.  This document aims to reduce the ambiguity by clarifying that only UPPERCASE usage of the key words have the  defined special meanings.</t></abstract>
</front>
<seriesInfo name='BCP' value='14'/>
<seriesInfo name='RFC' value='8174'/>
<seriesInfo name='DOI' value='10.17487/RFC8174'/>
</reference>

<reference anchor='RFC8866' target='https://www.rfc-editor.org/info/rfc8866'>
<front>
<title>SDP: Session Description Protocol</title>
<author fullname='A. Begen' initials='A.' surname='Begen'><organization/></author>
<author fullname='P. Kyzivat' initials='P.' surname='Kyzivat'><organization/></author>
<author fullname='C. Perkins' initials='C.' surname='Perkins'><organization/></author>
<author fullname='M. Handley' initials='M.' surname='Handley'><organization/></author>
<date month='January' year='2021'/>
<abstract><t>This memo defines the Session Description Protocol (SDP). SDP is intended for describing multimedia sessions for the purposes of session announcement, session invitation, and other forms of multimedia session initiation. This document obsoletes RFC 4566.</t></abstract>
</front>
<seriesInfo name='RFC' value='8866'/>
<seriesInfo name='DOI' value='10.17487/RFC8866'/>
</reference> website.
</t>
    </section>
  </middle>
  <back>
    <references>
      <name>References</name>
      <references>
        <name>Normative References</name>
        <xi:include href="https://bib.ietf.org/public/rfc/bibxml/reference.RFC.2119.xml"/>
        <xi:include href="https://bib.ietf.org/public/rfc/bibxml/reference.RFC.2736.xml"/>
        <xi:include href="https://bib.ietf.org/public/rfc/bibxml/reference.RFC.3264.xml"/>
        <xi:include href="https://bib.ietf.org/public/rfc/bibxml/reference.RFC.3550.xml"/>
        <xi:include href="https://bib.ietf.org/public/rfc/bibxml/reference.RFC.3551.xml"/>
        <xi:include href="https://bib.ietf.org/public/rfc/bibxml/reference.RFC.3711.xml"/>
        <xi:include href="https://bib.ietf.org/public/rfc/bibxml/reference.RFC.4585.xml"/>
        <xi:include href="https://bib.ietf.org/public/rfc/bibxml/reference.RFC.5124.xml"/>
        <xi:include href="https://bib.ietf.org/public/rfc/bibxml/reference.RFC.8174.xml"/>
        <xi:include href="https://bib.ietf.org/public/rfc/bibxml/reference.RFC.8866.xml"/>
      </references>

    <references title='Informative References'>

<reference anchor="AUDIOSCIP" target="https://www.iana.org/assignments/media-types/audio/scip">
  <front>
    <title>audio/scip: Internet Assigned Numbers Authority (IANA)</title>
    <author initials="M." surname="Faller">
      <organization></organization>
    </author>
    <author initials="D." surname="Hanson">
      <organization></organization>
    </author>
    <date year="2021" month="January" day="28"/>
  </front>
</reference>

<reference anchor='RFC4040' target='https://www.rfc-editor.org/info/rfc4040'>
<front>
<title>RTP Payload Format for a 64 kbit/s Transparent Call</title>
<author fullname='R. Kreuter' initials='R.' surname='Kreuter'><organization/></author>
<date month='April' year='2005'/>
<abstract><t>This document describes how to carry 64 kbit/s channel data transparently in RTP packets, using a pseudo-codec called &quot;Clearmode&quot;.  It also serves as registration for a related MIME type called &quot;audio/clearmode&quot;.</t><t>&quot;Clearmode&quot; is a basic feature of VoIP Media Gateways.  [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name='RFC' value='4040'/>
<seriesInfo name='DOI' value='10.17487/RFC4040'/>
</reference>
      <references>
        <name>Informative References</name>

        <reference anchor='RFC4855' target='https://www.rfc-editor.org/info/rfc4855'> anchor="MediaTypes" target="https://www.iana.org/assignments/media-types">
          <front>
            <title>Media Type Registration of RTP Payload Formats</title>
<author fullname='S. Casner' initials='S.' surname='Casner'><organization/></author>
<date month='February' year='2007'/>
<abstract><t>This document specifies the procedure to register RTP payload formats as audio, video, or other media subtype names.  This is useful in a text-based format description or control protocol to identify the type of an RTP transmission.  [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name='RFC' value='4855'/>
<seriesInfo name='DOI' value='10.17487/RFC4855'/>
</reference>

<reference anchor="RFC4961" target="https://www.rfc-editor.org/info/rfc4961">
<front>
<title>Symmetric RTP / RTP Control Protocol (RTCP)</title>
<author fullname="D. Wing" initials="D." surname="Wing"/>
<date month="July" year="2007"/>
<abstract>
<t>This document recommends using one UDP port pair for both communication directions of bidirectional RTP and RTP Control Protocol (RTCP) sessions, commonly called "symmetric RTP" and "symmetric RTCP". This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.</t>
</abstract>
</front>
<seriesInfo name="BCP" value="131"/>
<seriesInfo name="RFC" value="4961"/>
<seriesInfo name="DOI" value="10.17487/RFC4961"/>
</reference>

<reference anchor="RFC5109" target="https://www.rfc-editor.org/info/rfc5109">
<front>
<title>RTP Payload Format for Generic Forward Error Correction</title>
<author fullname="A. Li" initials="A." role="editor" surname="Li"/>
<date month="December" year="2007"/>
<abstract>
<t>This document specifies a payload format for generic Forward Error Correction (FEC) for media data encapsulated in RTP. It is based on the exclusive-or (parity) operation. The payload format described in this document allows end systems to apply protection using various protection lengths and levels, in addition to using various protection group sizes to adapt to different media and channel characteristics. It enables complete recovery of the protected packets or partial recovery of the critical parts of the payload depending on the packet loss situation. This scheme is completely compatible with non-FEC-capable hosts, so the receivers in a multicast group that do not implement FEC can still work by simply ignoring the protection data. This specification obsoletes RFC 2733 and RFC 3009. The FEC specified in this document is not backward compatible with RFC 2733 and RFC 3009. [STANDARDS-TRACK]</t>
</abstract>
</front>
<seriesInfo name="RFC" value="5109"/>
<seriesInfo name="DOI" value="10.17487/RFC5109"/>
</reference>

<reference anchor="RFC5761" target="https://www.rfc-editor.org/info/rfc5761">
<front>
<title>Multiplexing RTP Data and Control Packets on a Single Port</title>
<author fullname="C. Perkins" initials="C." surname="Perkins"/>
<author fullname="M. Westerlund" initials="M." surname="Westerlund"/>
<date month="April" year="2010"/>
<abstract>
<t>This memo discusses issues that arise when multiplexing RTP data packets and RTP Control Protocol (RTCP) packets on a single UDP port. It updates RFC 3550 and RFC 3551 to describe when such multiplexing is and is not appropriate, and it explains how the Session Description Protocol (SDP) can be used to signal multiplexed sessions. [STANDARDS-TRACK]</t>
</abstract>
</front>
<seriesInfo name="RFC" value="5761"/>
<seriesInfo name="DOI" value="10.17487/RFC5761"/>
</reference>

<reference anchor='RFC6184' target='https://www.rfc-editor.org/info/rfc6184'>
<front>
<title>RTP Payload Format for H.264 Video</title>
<author fullname='Y.-K. Wang' initials='Y.-K.' surname='Wang'><organization/></author>
<author fullname='R. Even' initials='R.' surname='Even'><organization/></author>
<author fullname='T. Kristensen' initials='T.' surname='Kristensen'><organization/></author>
<author fullname='R. Jesup' initials='R.' surname='Jesup'><organization/></author>
<date month='May' year='2011'/>
<abstract><t>This memo describes an RTP Payload format for the ITU-T Recommendation H.264 video codec and the technically identical ISO/IEC International Standard 14496-10 video codec, excluding the Scalable Video Coding (SVC) extension and the Multiview Video Coding extension, for which the RTP payload formats are defined elsewhere. The RTP payload format allows for packetization of one or more Network Abstraction Layer Units (NALUs), produced by an H.264 video encoder, in each RTP payload.  The payload format has wide applicability, as it supports applications from simple low bitrate conversational usage, to Internet video streaming with interleaved transmission, to high bitrate video-on-demand.</t><t>This memo obsoletes RFC 3984.  Changes from RFC 3984 are summarized in Section 14.  Issues on backward compatibility to RFC 3984 are discussed in Section 15.  [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name='RFC' value='6184'/>
<seriesInfo name='DOI' value='10.17487/RFC6184'/>
</reference>

<reference anchor='RFC6838' target='https://www.rfc-editor.org/info/rfc6838'>
<front>
<title>Media Type Specifications and Registration Procedures</title>
<author fullname='N. Freed' initials='N.' surname='Freed'><organization/></author>
<author fullname='J. Klensin' initials='J.' surname='Klensin'><organization/></author>
<author fullname='T. Hansen' initials='T.' surname='Hansen'><organization/></author>
<date month='January' year='2013'/>
<abstract><t>This document defines procedures for the specification and registration of media types for use in HTTP, MIME, and other Internet protocols.  This memo documents an Internet Best Current Practice.</t></abstract>
</front>
<seriesInfo name='BCP' value='13'/>
<seriesInfo name='RFC' value='6838'/>
<seriesInfo name='DOI' value='10.17487/RFC6838'/>
</reference>

<reference anchor='RFC7201' target='https://www.rfc-editor.org/info/rfc7201'>
<front>
<title>Options for Securing RTP Sessions</title>
<author fullname='M. Westerlund' initials='M.' surname='Westerlund'><organization/></author>
<author fullname='C. Perkins' initials='C.' surname='Perkins'><organization/></author>
<date month='April' year='2014'/>
<abstract><t>The Real-time Transport Protocol (RTP) is used in a large number of different application domains and environments.  This heterogeneity implies that different security mechanisms are needed to provide services such as confidentiality, integrity, and source authentication of RTP and RTP Control Protocol (RTCP) packets suitable for the various environments.  The range of solutions makes it difficult for RTP-based application developers to pick the most suitable mechanism.  This document provides an overview of a number of security solutions for RTP and gives guidance for developers on how to choose the appropriate security mechanism.</t></abstract>
</front>
<seriesInfo name='RFC' value='7201'/>
<seriesInfo name='DOI' value='10.17487/RFC7201'/>
</reference>

<reference anchor='RFC7202' target='https://www.rfc-editor.org/info/rfc7202'>
<front>
<title>Securing the RTP Framework: Why RTP Does Not Mandate a Single Media Security Solution</title>
<author fullname='C. Perkins' initials='C.' surname='Perkins'><organization/></author>
<author fullname='M. Westerlund' initials='M.' surname='Westerlund'><organization/></author>
<date month='April' year='2014'/>
<abstract><t>This memo discusses the problem of securing real-time multimedia sessions.  It also explains why the Real-time Transport Protocol (RTP) and the associated RTP Control Protocol (RTCP) do not mandate a single media security mechanism.  This is relevant for designers and reviewers of future RTP extensions to ensure that appropriate security mechanisms are mandated and that any such mechanisms are specified in a manner that conforms with the RTP architecture.</t></abstract>
</front>
<seriesInfo name='RFC' value='7202'/>
<seriesInfo name='DOI' value='10.17487/RFC7202'/>
</reference>

<reference anchor="RFC8083" target="https://www.rfc-editor.org/info/rfc8083">
<front>
<title>Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions</title>
<author fullname="C. Perkins" initials="C." surname="Perkins"/>
<author fullname="V. Singh" initials="V." surname="Singh"/>
<date month="March" year="2017"/>
<abstract>
<t>The Real-time Transport Protocol (RTP) is widely used in telephony, video conferencing, and telepresence applications. Such applications are often run on best-effort UDP/IP networks. If congestion control is not implemented in these applications, then network congestion can lead to uncontrolled packet loss and a resulting deterioration of the user's multimedia experience. The congestion control algorithm acts as a safety measure by stopping RTP flows from using excessive resources and protecting the network from overload. At the time of this writing, however, while there are several proprietary solutions, there is no standard algorithm for congestion control of interactive RTP flows.</t>
<t>This document does not propose a congestion control algorithm. It instead defines a minimal set of RTP circuit breakers: conditions under which an RTP sender needs to stop transmitting media data to protect the network from excessive congestion. It is expected that, in the absence of long-lived excessive congestion, RTP applications running on best-effort IP networks will be able to operate without triggering these circuit breakers. To avoid triggering the RTP circuit breaker, any Standards Track congestion control algorithms defined for RTP will need to operate within the envelope set by these RTP circuit breaker algorithms.</t>
</abstract>
</front>
<seriesInfo name="RFC" value="8083"/>
<seriesInfo name="DOI" value="10.17487/RFC8083"/>
<format target="https://www.rfc-editor.org/info/rfc8083" type="TXT"/>
</reference>

<reference anchor="RFC8085" target="https://www.rfc-editor.org/info/rfc8085">
<front>
<title>UDP Usage Guidelines</title>
<author fullname="L. Eggert" initials="L." surname="Eggert"/>
<author fullname="G. Fairhurst" initials="G." surname="Fairhurst"/>
<author fullname="G. Shepherd" initials="G." surname="Shepherd"/>
<date month="March" year="2017"/>
<abstract>
<t>The User Datagram Protocol (UDP) provides a minimal message-passing transport that has no inherent congestion control mechanisms. This document provides guidelines on the use of UDP for the designers of applications, tunnels, and other protocols that use UDP. Congestion control guidelines are a primary focus, but the document also provides guidance on other topics, including message sizes, reliability, checksums, middlebox traversal, the use of Explicit Congestion Notification (ECN), Differentiated Services Code Points (DSCPs), and ports.</t>
<t>Because congestion control is critical to the stable operation of the Internet, applications and other protocols that choose to use UDP as an Internet transport must employ mechanisms to prevent congestion collapse and to establish some degree of fairness with concurrent traffic. They may also need to implement additional mechanisms, depending on how they use UDP.</t>
<t>Some guidance is also applicable to the design of other protocols (e.g., protocols layered directly on IP or via IP-based tunnels), especially when these protocols do not themselves provide congestion control.</t>
<t>This document obsoletes RFC 5405 and adds guidelines for multicast UDP usage.</t>
</abstract>
</front>
<seriesInfo name="BCP" value="145"/>
<seriesInfo name="RFC" value="8085"/>
<seriesInfo name="DOI" value="10.17487/RFC8085"/>
<format target="https://www.rfc-editor.org/info/rfc8085" type="TXT"/>
</reference>

<reference anchor='RFC8088' target='https://www.rfc-editor.org/info/rfc8088'>
<front>
<title>How to Write an RTP Payload Format</title>
<author fullname='M. Westerlund' initials='M.' surname='Westerlund'><organization/></author>
<date month='May' year='2017'/>
<abstract><t>This document contains information on how best to write an RTP payload format specification.  It provides reading tips, design practices, and practical tips on how to produce an RTP payload format specification quickly and with good results.  A template is also included with instructions.</t></abstract>
</front>
<seriesInfo name='RFC' value='8088'/>
<seriesInfo name='DOI' value='10.17487/RFC8088'/>
</reference>

<reference anchor='RFC8130' target='https://www.rfc-editor.org/info/rfc8130'>
<front>
<title>RTP Payload Format for the Mixed Excitation Linear Prediction Enhanced (MELPe) Codec</title>
<author fullname='V. Demjanenko' initials='V.' surname='Demjanenko'><organization/></author>
<author fullname='D. Satterlee' initials='D.' surname='Satterlee'><organization/></author>
<date month='March' year='2017'/>
<abstract><t>This document describes the RTP payload format for the Mixed Excitation Linear Prediction Enhanced (MELPe) speech coder.  MELPe's three different speech encoding rates and sample frame sizes are supported.  Comfort noise procedures and packet loss concealment are described in detail.</t></abstract>
</front>
<seriesInfo name='RFC' value='8130'/>
<seriesInfo name='DOI' value='10.17487/RFC8130'/>
</reference>

<reference anchor="RFC9143" target="https://www.rfc-editor.org/info/rfc9143">
<front>
<title>Negotiating Media Multiplexing Using the Session Description Protocol (SDP)</title>
<author fullname="C. Holmberg" initials="C." surname="Holmberg"/>
<author fullname="H. Alvestrand" initials="H." surname="Alvestrand"/>
<author fullname="C. Jennings" initials="C." surname="Jennings"/>
<date month="February" year="2022"/>
<abstract>
<t>This specification defines a new Session Description Protocol (SDP) Grouping Framework extension called 'BUNDLE'. The extension can be used with the SDP offer/answer mechanism to negotiate the usage of a single transport (5-tuple) for sending and receiving media described by multiple SDP media descriptions ("m=" sections). Such transport is referred to as a "BUNDLE transport", and the media is referred to as "bundled media". The "m=" sections that use the BUNDLE transport form a BUNDLE group.</t>
<t>This specification defines a new RTP Control Protocol (RTCP) Source Description (SDES) item and a new RTP header extension.</t>
<t>This specification updates RFCs 3264, 5888, and 7941.</t>
<t>This specification obsoletes RFC 8843.</t>
</abstract>
</front>
<seriesInfo name="RFC" value="9143"/>
<seriesInfo name="DOI" value="10.17487/RFC9143"/>
</reference>

<reference anchor="RFC9170" target="https://www.rfc-editor.org/info/rfc9170">
<front>
<title>Long-Term Viability of Protocol Extension Mechanisms</title>
<author fullname="M. Thomson" initials="M." surname="Thomson"/>
<author fullname="T. Pauly" initials="T." surname="Pauly"/>
<date month="December" year="2021"/>
<abstract>
<t>The ability to change protocols depends on exercising the extension and version-negotiation mechanisms that support change. This document explores how regular use of new protocol features can ensure that it remains possible to deploy changes to a protocol. Examples are given where lack of use caused changes to be more difficult or costly.</t>
</abstract> Types</title>
            <author>
	      <organization>IANA</organization>
            </author>
          </front>
<seriesInfo name="RFC" value="9170"/>
<seriesInfo name="DOI" value="10.17487/RFC9170"/>
        </reference>

        <xi:include href="https://bib.ietf.org/public/rfc/bibxml/reference.RFC.4040.xml"/>
        <xi:include href="https://bib.ietf.org/public/rfc/bibxml/reference.RFC.4855.xml"/>
        <xi:include href="https://bib.ietf.org/public/rfc/bibxml/reference.RFC.4961.xml"/>
        <xi:include href="https://bib.ietf.org/public/rfc/bibxml/reference.RFC.5109.xml"/>
        <xi:include href="https://bib.ietf.org/public/rfc/bibxml/reference.RFC.5761.xml"/>
        <xi:include href="https://bib.ietf.org/public/rfc/bibxml/reference.RFC.6184.xml"/>
        <xi:include href="https://bib.ietf.org/public/rfc/bibxml/reference.RFC.6838.xml"/>
        <xi:include href="https://bib.ietf.org/public/rfc/bibxml/reference.RFC.7201.xml"/>
        <xi:include href="https://bib.ietf.org/public/rfc/bibxml/reference.RFC.7202.xml"/>
        <xi:include href="https://bib.ietf.org/public/rfc/bibxml/reference.RFC.8083.xml"/>
        <xi:include href="https://bib.ietf.org/public/rfc/bibxml/reference.RFC.8085.xml"/>
        <xi:include href="https://bib.ietf.org/public/rfc/bibxml/reference.RFC.8088.xml"/>
        <xi:include href="https://bib.ietf.org/public/rfc/bibxml/reference.RFC.8130.xml"/>
        <xi:include href="https://bib.ietf.org/public/rfc/bibxml/reference.RFC.9143.xml"/>
        <xi:include href="https://bib.ietf.org/public/rfc/bibxml/reference.RFC.9170.xml"/>

        <reference anchor="RMCAT" target="https://datatracker.ietf.org/wg/rmcat/about/" quoteTitle="true" derivedAnchor="RMCAT"> target="https://datatracker.ietf.org/wg/rmcat/about">
          <front>
            <title>RTP Media Congestion Avoidance Techniques (rmcat) Working Group</title> (rmcat)</title>
            <author>
<organization showOnFrontPage="true">IETF</organization>
              <organization>IETF</organization>
            </author>
          </front>
        </reference>

        <reference anchor="SCIP210" target='https://www.iad.gov/SecurePhone/index.cfm'> anchor="SCIP210">
          <front>
            <title>SCIP Signaling Plan</title> Plan, SCIP-210</title>
            <author>
              <organization>SCIP Working Group</organization>
            </author>
    <date year="2023" month="September"/>
  </front>
  <refcontent>SCIP-210, r3.11</refcontent>
</reference>

<reference anchor="VIDEOSCIP" target="https://www.iana.org/assignments/media-types/video/scip">
  <front>
    <title>video/scip: Internet Assigned Numbers Authority (IANA)</title>
    <author initials="M." surname="Faller">
      <organization></organization>
    </author>
    <author initials="D." surname="Hanson">
      <organization></organization>
    </author>
    <date year="2021" month="January" day="28"/>
          </front>
	  <annotation>Available by request via email to &lt;ncia.cis3@ncia.nato.int&gt;.</annotation>
       </reference>

    </references>
    </references>
  </back>

</rfc>