<?xmlversion="1.0" encoding="UTF-8"?> <?xml-stylesheet type="text/xsl" href="rfc2629.xslt" ?> <!-- generated by https://github.com/cabo/kramdown-rfc version 1.6.15 (Ruby 3.0.2) -->version='1.0' encoding='UTF-8'?> <!DOCTYPE rfc [ <!ENTITY nbsp " "> <!ENTITY zwsp "​"> <!ENTITY nbhy "‑"> <!ENTITY wj "⁠"> ]><?rfc {"toc"=>nil, "sortrefs"=>nil, "symrefs"=>nil}="yes"?><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> <!--[rfced] Please insert any keywords (beyond those that appear in the title) for use on https://www.rfc-editor.org/search. --> <keyword>example</keyword> <!-- [rfced] Because of the abstract's length, we suggest moving some of its content to the Introduction. As noted in Section 4.3 of RFC 7322: Every RFC must have an Abstract that provides a concise and comprehensive overview of the purpose and contents of the entire document, to give a technically knowledgeable reader a general overview of the function of the document.... A satisfactory Abstract can often be constructed in part from material within the Introduction section, but an effective Abstract may be shorter, less detailed, and perhaps broader in scope than the Introduction. Please let us know how the text may be updated. --> <abstract> <t>This document describes the RTP payload format of the Secure Communication Interoperability Protocol (SCIP). SCIP is anapplication layerapplication-layer protocol that provides end-to-end capability exchange, packetization/de-packetization of media, reliable transport, and payload encryption.</t> <t>SCIP handles packetization/de-packetization oftheencrypted media and acts as a tunneling protocol, treating SCIP payloads as opaque octets to be encapsulated within RTP payloads prior to transmission or decapsulated on reception. SCIP payloads are sized to fit within the maximum transmission unit (MTU) when transported overRTPRTP, thereby avoiding fragmentation.</t> <t>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.</t> <t>To establish reliable communications using SCIP over RTP, it is important thatmiddle boxesmiddleboxes avoid parsing or modifying SCIP payloads. Because SCIP payloads are confidentiality and integrity protected and are only decipherable by the originating 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 is parsing unnecessary to tunnel/detunnel SCIP within RTP, but the parsing and filtering of SCIP payloads by middle boxes would likely lead to ossification of the evolving SCIP protocol.</t> </abstract> <note> <name>IESG Note</name> <t>This IETF specification depends upon a second technical specification that is not available publicly, namely <xref target="SCIP210"/>. The IETF was therefore unable to conduct a security review of that specification, independently or when carried inside Audio/Video Transport (AVT). Implementers need to be aware that the IETF hence cannot verify any of the security claims contained in this document.</t> </note> </front> <middle> <sectionanchor="key-points"><name>Keyanchor="key-points"> <name>Key Points</name> <!-- section 1 --> <ul> <li>SCIP is anapplication layerapplication-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 <xreftarget="RFC4040"/> Clearmode.</li>target="RFC4040"/>.</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 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 to parse SCIP payloads. Since SCIP payloads are integrity protected, modification of the SCIP payload is detected as an integrity violation by SCIPendpointsendpoints, 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 usefulpurpose.</li></ul>purpose.</li> </ul> </section> <sectionanchor="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.TheThis 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 document details usage<!--[rfced] For ease of the"audio/scip" and "video/scip" pseudo-codecs <xref target="AUDIOSCIP"/>, <xref target="VIDEOSCIP"/>reader, we have formatted the text as asecure session establishment protocollist. In addition, we changed the verbs to make these list items more parallel andmedia transport protocol over RTP.consistent. Please review and let us know if this conveys the intended meaning. Original: It discusses (1) how encrypted audio and video codec payloads are transported over RTP; (2) the IP network layer not implementing SCIP as a protocol since SCIP operates at the application layer in endpoints; (3) the IP network layer enabling SCIP traffic to transparently pass through the network; (4) network devices not recognizing SCIP, and thus removing the scip codecs from the SDP media payload declaration before forwarding to the next network node; and finally, (5) SCIP endpoint devices not operating on networks due to the scip media subtype removal from the SDP media payload declaration.</t>Perhaps: It discusses how: 1. encrypted audio and video codec payloads are transported over RTP; 2. the IP network layer does not implement SCIP as a protocol since SCIP operates at the application layer in endpoints; 3. the IP network layer enables SCIP traffic to transparently pass through the network; 4. network devices do not recognize SCIP, and thus remove the SCIP codecs from the SDP media payload declaration before forwarding to the next network node; and finally, 5. SCIP endpoint devices do not operate on networks due to the scip media subtype removal from the SDP media payload declaration. --> <t>This document details usage of the "audio/scip" <xref target="AUDIOSCIP"/> and "video/scip" <xref target="VIDEOSCIP"/> pseudo-codecs as a secure session establishment protocol and media transport protocol over RTP. It discusses:</t> <ol spacing="normal"> <li>how encrypted audio and video codec payloads are transported over RTP;</li> <li>the IP network layer not implementing SCIP as a protocol since SCIP operates at the application layer in endpoints;</li> <li>the IP network layer enabling SCIP traffic to transparently pass through the network;</li> <li>network devices not recognizing SCIP, and thus removing the SCIP codecs from the SDP media payload declaration before forwarding to the next network node; and finally,</li> <li>SCIP endpoint devices not operating on networks due to the scip media subtype removal from the SDP media payload 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 andfrequencyfrequency, 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, retransmission, capability exchange, version negotiation, and payload encryption. Since the traffic is encrypted, neither the RTP transport nor middle boxes 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, 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> <sectionanchor="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 inBCP 14BCP 14 <xref target="RFC2119"/> <xref target="RFC8174"/> when, and only when, they appear in all capitals, as shownhere.</t> <t>Besthere. </t> <t>The best current practices for writing an RTP payload formatspecification were followedspecification, as per <xref target="RFC2736"/> and <xreftarget="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> <sectionanchor="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 Profile</dd> <dt>AVPF:</dt> <dd>Audio/Video Profile Feedback</dd> <dt>ICWG:</dt> <dd>Interoperability Control Working Group</dd> <dt>IICWG:</dt> <dd>International Interoperability Control Working Group</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 Profile</dd> <dt>SAVPF:</dt> <dd>Secure Audio/Video Profile 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 Transport Protocol</dd> <dt>STANAG:</dt> <dd>Standardization Agreement</dd> </dl> </section> </section> <sectionanchor="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, NATOpartnerspartners, 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 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 the audio/scip and video/scip media subtypes thatenablesenable 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 policywhichthat 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> <sectionanchor="media-format-description"><name>Payloadanchor="media-format-description"> <name>Payload Format</name> <!-- section 4 --> <t>The "scip" media subtype identifies and indicates support forand identifiesSCIP traffic that is being transported over RTP. Transcoding, lossy compression, or other data modificationsMUST NOT<bcp14>MUST NOT</bcp14> be performed by the network on the SCIP RTP payload. The audio/scip and 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 anapplication layerapplication-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> <figureanchor="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 | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ | | | SCIPpayloadPayload | | | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+</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 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 endpointsMAY<bcp14>MAY</bcp14> employ mechanisms, such asInter-mediainter-media RTPSynchronizationsynchronization as described in <xreftarget="RFC8088"/> Section 3.3.4,target="RFC8088" sectionFormat="comma" section="3.3.4"/>, to synchronize audio/scip and 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> <figureanchor="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*(<=(<= 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 andretransmission. </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 evolveandand, assuchsuch, the SCIP RTP trafficMUST 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 devicesMUST NOT<bcp14>MUST NOT</bcp14> modify SCIP RTP packets.</t> <sectionanchor="rtp-header-fields"><name>RTPanchor="rtp-header-fields"> <name>RTP Header Fields</name> <!-- section 4.1 --> <t>The SCIP RTP header fieldsSHALL<bcp14>SHALL</bcp14> conform toRFC 3550.</t><xref target="RFC3550"/>.</t> <t>SCIP traffic may be continuous or discontinuous. The Timestamp fieldMUST<bcp14>MUST</bcp14> increment based on the sampling clock for discontinuous transmission as described in <xreftarget="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 bitSHALL<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> <sectionanchor="congestion-control"><name>Congestionanchor="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 UDPSHOULD<bcp14>SHOULD</bcp14> implement their own congestion control above the UDP layer <xref target="RFC8085"/> andMAY<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"/> areOPTIONAL<bcp14>OPTIONAL</bcp14> because SCIP itself manages retransmissions of some errored or lost packets. Specifically, thePayload-Specific Feedback Messagespayload-specific feedback messages defined inRFC 4585 section 6.3<xref target="RFC4585" sectionFormat="comma" section="6.3"/> areOPTIONAL<bcp14>OPTIONAL</bcp14> when transporting video data. </t> </section> <sectionanchor="augmented-protocols"><name>Useanchor="augmented-protocols"> <name>Use of Augmented RTP Transport Protocols with SCIP</name> <!-- section 4.3 --> <!--[rfced] May the title of Section 4.3 be updated as follows, as this seems redundant (where "RTP" is Real-time Transport Protocol)? Original: 4.3. Use of Augmented RTP Transport Protocols with SCIP Perhaps: 4.3. Use of Augmented RTPs with SCIP Similarly, may it be updated in "Additional RTP transport protocols that..."? --> <!--[rfced] The list items below have been updated with quotes because they are the titles of RFCs. FYI, the fourth list item has been updated to match the title of RFC 9143 (where "a.k.a. BUNDLE" is afterwards). Please review. Original: Additional RTP transport protocols that do not modify the SCIP payload are considered OPTIONAL in this document and are discretionary for a SCIP device vendor to implement. Some examples include but are not limited to: * RTP Payload Format for Generic Forward Error Correction [RFC5109] * Multiplexing RTP Data and Control Packets on a Single Port [RFC5761] * Symmetric RTP/RTP Control Protocol (RTCP) [RFC4961] * Negotiating Media Multiplexing Using the Session Description Protocol (BUNDLE) [RFC9143] Current: Additional RTP transport protocols that do not modify the SCIP payload are considered OPTIONAL in this document and are discretionary for a SCIP device vendor to implement. Some examples include, but are not limited to: * "RTP Payload Format for Generic Forward Error Correction" [RFC5109] * "Multiplexing RTP Data and Control Packets on a Single Port" [RFC5761] * "Symmetric RTP / RTP Control Protocol (RTCP)" [RFC4961] * "Negotiating Media Multiplexing Using the Session Description Protocol (SDP)" a.k.a. BUNDLE [RFC9143] --> <t>The SCIPapplication layerapplication-layer protocol uses RTP as a basic transport for the audio/scip and video/scip payloads. AdditionalRTP transport protocolsRTPs that do not modify the SCIP payload are consideredOPTIONAL<bcp14>OPTIONAL</bcp14> in this document and are discretionary for a SCIP device vendor to implement. Some examplesincludeinclude, but are not limited to:</t> <ul><li>RTP<li>"RTP Payload Format for Generic Forward ErrorCorrectionCorrection" <xref target="RFC5109"/></li><li>Multiplexing<li>"Multiplexing RTP Data and Control Packets on a SinglePortPort" <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. BUNDLE <xref target="RFC9143"/></li> </ul> </section> </section> <sectionanchor="payload-format-parameters"><name>Payloadanchor="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 templateformfrom <xref target="RFC6838"/>. A clock rate of 8000 HzSHALL<bcp14>SHALL</bcp14> be used for "audio/scip". A clock rate of 90000 HzSHALL<bcp14>SHALL</bcp14> be used for "video/scip".</t> <sectionanchor="media-subtype-audioscip"><name>Mediaanchor="media-subtype-audioscip"> <!--[rfced] FYI, we made some formatting adjustments to the media type registrations in Sections 5.1 and 5.2 per the template specified in RFC 6838. In addition, we have updated the pointer to the security considerations (2 instances). Original (links to IANA Considerations): Security considerations: See Section 7. Current (links to Security Considerations): Security considerations: See Section 6. --> <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. ThereSHALL<bcp14>SHALL</bcp14> be no encoding/decoding (transcoding) of the audio stream as it traverses thenetwork.</t> <t>Security considerations: See Section 7.</t> <t>Interoperability considerations: N/A</t> <t>Published specifications: <xref target="SCIP210"/></t> <t>Applications whichnetwork.</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 thismedia: N/A</t> <t>Fragment Identifier considerations: none</t> <t>Restrictions on usage: N/A</t> <t>Additional information:</t> <t indent="3">1. Deprecatedmedia 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 thistype: 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. Macintoshtype:</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 typecode: N/A</t> <t indent="3">5. Object Identifiers: N/A</t> <t>Personcode(s):</dt><dd>N/A</dd> </dl> </dd> <dt>Person & email address to contact for furtherinformation:</t> <t indent="3">1. Name: Michaelinformation:</dt><dd>Michael Faller (michael.faller@gd-ms.com) and DanielHanson</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">MichaelHanson (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) 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> <sectionanchor="media-subtype-videoscip"><name>Mediaanchor="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. ThereSHALL<bcp14>SHALL</bcp14> be no encoding/decoding (transcoding) of the video stream as it traverses thenetwork.</t> <t>Security considerations: See Section 7.</t> <t>Interoperability considerations: N/A</t> <t>Published specifications: <xref target="SCIP210"/></t> <t>Applications whichnetwork.</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 thismedia: N/A</t> <t>Fragment Identifier considerations: none</t> <t>Restrictions on usage: N/A</t> <t>Additional information:</t> <t indent="3">1. Deprecatedmedia 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 thistype: 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. Macintoshtype:</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 typecode: N/A</t> <t indent="3">5. Object Identifiers: N/A</t> <t>Personcode(s):</dt><dd>N/A</dd> </dl> </dd> <dt>Person & email address to contact for furtherinformation:</t> <t indent="3">1. Name: Michaelinformation:</dt><dd>Michael Faller (michael.faller@gd-ms.com) and DanielHanson</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">MichaelHanson (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) 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> <sectionanchor="mapping-to-sdp"><name>Mappinganchor="mapping-to-sdp"> <name>Mapping to SDP</name> <!-- section 5.3 --> <t>The mapping of theabove definedabove-defined payload format media subtype and its parametersSHALL<bcp14>SHALL</bcp14> be implemented according toSection 3 of<xreftarget="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, and the media type ("video") goes in SDP "m=" as the media name for 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 is:</t><figure> <artwork> <![CDATA[<artwork><![CDATA[ m=audio 50000 RTP/AVP 96 a=rtpmap:96scip/8000]]> </artwork> </figure>scip/8000 ]]></artwork> <t>An example mapping for video/scip is:</t><figure> <artwork> <![CDATA[<artwork><![CDATA[ m=video 50002 RTP/AVP 97 a=rtpmap:97scip/90000]]> </artwork> </figure>scip/90000 ]]></artwork> <t>An example mapping for both audio/scip and video/scip is:</t><figure> <artwork> <![CDATA[<artwork><![CDATA[ m=audio 50000 RTP/AVP 96 a=rtpmap:96 scip/8000 m=video 50002 RTP/AVP 97 a=rtpmap:97scip/90000]]> </artwork> </figure>scip/90000 ]]></artwork> </section> <sectionanchor="sdp-offeranswer-considerations"><name>SDPanchor="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 deviceSHALL<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[<artwork><![CDATA[ m=audio 50000 RTP/AVP 96 0 8 a=rtpmap:96 scip/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8PCMA/8000]]> </artwork> </figure>PCMA/8000 ]]></artwork> </section> </section> <sectionanchor="security-considerations"><name>Securityanchor="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 RTPProtocolFramework: 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"/>. ApplicationsSHOULD<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 pathologicaldata. Nordata, 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 RTPheaderheaders and/or RTCP security might consider mechanisms such as SRTP <xref target="RFC3711"/>, however these additional mechanisms are consideredOPTIONAL<bcp14>OPTIONAL</bcp14> in this document.</t> </section> <sectionanchor="iana-considerations"><name>IANAanchor="iana-considerations"> <name>IANA Considerations</name> <!-- section 7 --> <t>The audio/scip and video/scip media subtypes have previously been registered with IANA <xref target="AUDIOSCIP"/> <xref target="VIDEOSCIP"/>. IANA should update <xref target="AUDIOSCIP"/> and <xref target="VIDEOSCIP"/> to reference this document upon publication.</t> </section> <!--[rfced] Does the "IANA should update..." sentence below request that IANA A) update the references within the registrations (which are in Sections 5.1 and 5.2) to point to this RFC rather than [SCIP210], or B) update the Reference column in the media type registry (https://www.iana.org/assignments/media-types) Original: The audio/scip and video/scip media subtypes have previously been registered with IANA [AUDIOSCIP] [VIDEOSCIP]. IANA should update [AUDIOSCIP] and [VIDEOSCIP] to reference this document upon publication. Assuming (B) was your intention, we suggest the following update. Please let us know if you agree. Suggested: The audio/scip and video/scip media subtypes have previously been registered in the "Media Types" registry [MediaTypes]. IANA has updated these registrations to reference this document. In addition, Section 2 would be updated accordingly: Original: The document details usage of the "audio/scip" and "video/scip" pseudo-codecs [AUDIOSCIP], [VIDEOSCIP] as a secure session establishment protocol and media transport protocol over RTP. Suggested: The document details usage of the "audio/scip" and "video/scip" pseudo-codecs [MediaTypes] as a secure session establishment protocol and media transport protocol over RTP. where the reference is as follows: [MediaTypes] IANA, "Media Types", <https://www.iana.org/assignments/media-types>. --> <sectionanchor="scip-contact-info"><name>SCIPanchor="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 may be requested from the email address below. </t><t> SCIP<contact> <organization>SCIP Working Group, CIS3Partnership<br/> NATOPartnership</organization> <address> <postal> <postalLine>NATO Communications and InformationAgency<br/> OudeAgency</postalLine> <postalLine>Oude Waalsdorperweg61<br/> 259761</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'><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> <name>Informative References</name> <referenceanchor='RFC2119' target='https://www.rfc-editor.org/info/rfc2119'>anchor="AUDIOSCIP" target="https://www.iana.org/assignments/media-types/audio/scip"> <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><title>audio/scip</title> <author> <organization>IANA</organization> </author> </front><seriesInfo name='BCP' value='14'/> <seriesInfo name='RFC' value='2119'/> <seriesInfo name='DOI' value='10.17487/RFC2119'/></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"/> <referenceanchor='RFC2736' target='https://www.rfc-editor.org/info/rfc2736'>anchor="RMCAT" target="https://datatracker.ietf.org/wg/rmcat/about"> <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><title>RTP Media Congestion Avoidance Techniques (rmcat)</title> <author> <organization>IETF</organization> </author> </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.<!--[rfced] Theoffer/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 deliveryURL provided ina 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 ofthe textin 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 rulesandalgorithms governing howreference below gives theprotocol is used. The biggest changemessage "Warning: Potential Security Risk Ahead" (Firefox) and "Your connection is not private" (Chrome). Is there anenhancement 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 "RTP/AVP"alternative URL available forthe use of the real-time transport protocol (RTP),this resource? Original (in Section 8, SCIP Contact Information): An older public version2, and the associated control protocol, RTCP, within audio and video multiparticipant conferences with minimal control. It provides interpretationsofgeneric fields withintheRTPSCIP-210 specificationsuitable for audio and video conferences. In particular, this document defines a set of default mappings from payload type numbers to encodings. This document also describes how audio and video data maycan becarried within RTP. It defines a set of standard encodings and their names when used within RTP. The descriptions provide pointers 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 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>downloaded from https://www.iad.gov/SecurePhone/index.cfm. Original (in Section 9.2, Informative References): [SCIP210] SCIP Working Group, "SCIP Signaling Plan", SCIP-210, r3.11, September 2023, <https://www.iad.gov/SecurePhone/index.cfm>. --> <referenceanchor='RFC3711' target='https://www.rfc-editor.org/info/rfc3711'>anchor="SCIP210" target="https://www.iad.gov/SecurePhone/index.cfm"> <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><title>SCIP Signaling Plan</title> <author> <organization>SCIP Working Group</organization> </author> <datemonth='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>year="2023" month="September"/> </front><seriesInfo name='RFC' value='3711'/> <seriesInfo name='DOI' value='10.17487/RFC3711'/><refcontent>SCIP-210, r3.11</refcontent> </reference> <referenceanchor='RFC4585' target='https://www.rfc-editor.org/info/rfc4585'>anchor="VIDEOSCIP" target="https://www.iana.org/assignments/media-types/video/scip"> <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><title>video/scip</title> <author> <organization>IANA</organization> </author> </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</references> </references> </back> <!--[rfced] Please review thereceivers to a sender are defined in RFC 3711following questions andRFC 4585, respectively. This memo specifieschanges regarding thecombination 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 beterminology used inprotocol 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 forthis document. a. We note thepurposesuse ofsession announcement, session invitation,scip, audio/scip, andother 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> </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>Thisvideo/scip with and without quotes throughout this documentdescribes how to carry 64 kbit/s channel data transparently(as seen inRTP packets, using a pseudo-codec called "Clearmode". It also serves as registration for a related MIME type called "audio/clearmode".</t><t>"Clearmode" 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> <reference anchor='RFC4855' target='https://www.rfc-editor.org/info/rfc4855'> <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 specifiestheprocedure to register RTP payload formats as audio, video, or other media subtype names. This is useful in a text-based format description or control protocolexample below). How would you like these terms toidentify 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 pairappear forboth communication directions of bidirectional RTP and RTP Control Protocol (RTCP) sessions, commonly called "symmetric RTP" and "symmetric RTCP".consistency? Original: This documentspecifies an Internet Best Current Practices for the Internet Community, and requests discussionprovides essential information about audio/scip andsuggestions 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 Formatvideo/scip media subtypes that enables network equipment manufacturers to include settings forGeneric Forward Error Correction</title> <author fullname="A. Li" initials="A." role="editor" surname="Li"/> <date month="December" year="2007"/> <abstract> <t>This document specifies"scip" as apayload format for generic Forward Error Correction (FEC) forknown audio and video mediadata encapsulated in RTP. It is based on the exclusive-or (parity) operation. The payload format describedsubtype inthis document allows end systemstheir equipment. This enables network administrators toapply protection using various protection lengthsdefine andlevels, in addition to using various protection group sizes to adapt to differentimplement a compatible security policy which includes audio and video media subtypes "audio/scip" andchannel characteristics. It enables complete recovery of the protected packets or partial recovery of the critical parts of the payload depending"video/scip", respectively, as permitted codecs on thepacket loss situation. This scheme is completely compatible with non-FEC-capable hosts, sonetwork. b. Please review usage of thereceivers in a multicast group that do not implement FEC can still work by simply ignoring"/" character throughout. In theprotection 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, andinstances below, itexplains how the Session Description Protocol (SDP) canmay beusedunclear tosignal 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 whicha reader whether theRTP payload formats are defined elsewhere. The RTP payload format allows"/" stands forpacketization of one"and" ormore 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"or". For clarity, we recommend updating these instances toRFC 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)"and", "or", or "and/or". Originals: SCIP isused in a large number of differentan applicationdomains and environments. This heterogeneity implieslayer protocol thatdifferent 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 documentprovidesan overview of a numberend-to-end capability exchange, packetization/de-packetization ofsecurity 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 Doesmedia... NotMandate 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. Thisonly isrelevant for designers and reviewers of future RTP extensionsparsing unnecessary toensure 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 applicationstunnel/detunnel SCIP within RTP... SCIP standards areoften run on best-effort UDP/IP networks. If congestion control is not implemented in these applications, then network congestion can leadcurrently available touncontrolled packet loss and a resulting deteriorationparticipating government/ military communities... There SHALL be no encoding/decoding (transcoding) of theuser's multimedia experience.audio stream as it traverses the network. c. FYI - Thecongestion control algorithm actsterms below have been updated asa safety measure by stopping RTP flows from using excessive resourcesfollows. Please review andprotecting 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 oflet us know if you have any additional changes. Inter-media RTPcircuit breakers: conditions under which anSynchronization -> inter-media RTPsender needs to stop transmitting media datasynchronization toprotect the network from excessive congestion. Itmatch RFC 8088, which isexpected that, in the absence of long-lived excessive congestion, RTP applications running on best-effort IP networks will be ablecited. Payload-Specific Feedback Messages -> payload-specific feedback messages tooperate without triggering these circuit breakers. To avoid triggering the RTP circuit breaker, any Standards Track congestion control algorithms definedmatch RFC 4585, which is cited. scip codecs -> SCIP codecs forRTP will need to operateconsistency with other instances withinthe 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 UDPthis document. --> <!-- [rfced] FYI - We have added expansions forthe 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 theyabbreviations upon first useUDP.</t> <t>Some guidance is also applicable to the designper Section 3.6 ofother 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 obsoletesRFC5405 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>This7322 ("RFC Style Guide"). Please review each expansion in the documentcontains information on how bestcarefully towrite an RTP payload format specification. It provides reading tips, design practices, and practical tips on howensure correctness. MELPe expanded toproduce 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 theMixed 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--> <!--[rfced] Please review each artwork element indetail.</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 UsingtheSession 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 canXML file. Specifically, should any artwork element in Sections 5.3 and 5.4 be tagged as sourcecode or another element? (We note that RFC 9143 usedwith the SDP offer/answer mechanism to negotiatetheusage of a single transport (5-tuple)sourcecode element with type="sdp" forsending 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 usewhat may be similar text.) Relevant documentation: https://authors.ietf.org/rfcxml-vocabulary#artwork https://authors.ietf.org/rfcxml-vocabulary#sourcecode --> <!--[rfced] Please review theBUNDLE 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"Inclusive Language" portion ofProtocol 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 exercisingtheextension and version-negotiation mechanisms that support change. This document explores how regular use of new protocol features can ensure that it remains possible to deployonline Style Guide <https://www.rfc-editor.org/styleguide/part2/#inclusive_language> and let us know if any changesto a protocol. Examplesaregiven where lack of use caused changes toneeded. Note that our script did not flag any words in particular, but this should still bemore difficult or costly.</t> </abstract> </front> <seriesInfo name="RFC" value="9170"/> <seriesInfo name="DOI" value="10.17487/RFC9170"/> </reference> <reference anchor="RMCAT" target="https://datatracker.ietf.org/wg/rmcat/about/" quoteTitle="true" derivedAnchor="RMCAT"> <front> <title>RTP Media Congestion Avoidance Techniques (rmcat) Working Group</title> <author> <organization showOnFrontPage="true">IETF</organization> </author> </front> </reference> <reference anchor="SCIP210" target='https://www.iad.gov/SecurePhone/index.cfm'> <front> <title>SCIP Signaling Plan</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> </reference> </references> </back>reviewed as a best practice. --> </rfc>