<pre>Internet Engineering Task Force (IETF)                        L. Portman
Request for Comments: 7866                                  NICE Systems
Category: Standards Track                                    H. Lum, Ed.
ISSN: 2070-1721                                                  Genesys
                                                                C. Eckel
                                                                   Cisco
                                                             A. Johnston
                                        Illinois Institute of Technology
                                                               A. Hutton
                                                                   Unify
                                                                May 2016


                       <span class="h1">Session Recording Protocol</span>

Abstract

   This document specifies the use of the Session Initiation Protocol
   (SIP), the Session Description Protocol (SDP), and the Real-time
   Transport Protocol (RTP) for delivering real-time media and metadata
   from a Communication Session (CS) to a recording device.  The Session
   Recording Protocol specifies the use of SIP, SDP, and RTP to
   establish a Recording Session (RS) between the Session Recording
   Client (SRC), which is on the path of the CS, and a Session Recording
   Server (SRS) at the recording device.  This document considers only
   active recording, where the SRC purposefully streams media to an SRS
   and all participating user agents (UAs) are notified of the
   recording.  Passive recording, where a recording device detects media
   directly from the network (e.g., using port-mirroring techniques), is
   outside the scope of this document.  In addition, lawful intercept is
   outside the scope of this document.

Status of This Memo

   This is an Internet Standards Track document.

   This document is a product of the Internet Engineering Task Force
   (IETF).  It represents the consensus of the IETF community.  It has
   received public review and has been approved for publication by the
   Internet Engineering Steering Group (IESG).  Further information on
   Internet Standards is available in <a href="./rfc7841#section-2">Section&nbsp;2 of RFC 7841</a>.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at
   <a href="http://www.rfc-editor.org/info/rfc7866">http://www.rfc-editor.org/info/rfc7866</a>.






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Copyright Notice

   Copyright (c) 2016 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to <a href="https://www.rfc-editor.org/bcp/bcp78">BCP 78</a> and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (<a href="http://trustee.ietf.org/license-info">http://trustee.ietf.org/license-info</a>) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   <a href="#section-1">1</a>. Introduction ....................................................<a href="#page-4">4</a>
   <a href="#section-2">2</a>. Terminology .....................................................<a href="#page-4">4</a>
   <a href="#section-3">3</a>. Definitions .....................................................<a href="#page-4">4</a>
   <a href="#section-4">4</a>. Scope ...........................................................<a href="#page-4">4</a>
   <a href="#section-5">5</a>. Overview of Operations ..........................................<a href="#page-5">5</a>
      <a href="#section-5.1">5.1</a>. Delivering Recorded Media ..................................<a href="#page-5">5</a>
      <a href="#section-5.2">5.2</a>. Delivering Recording Metadata ..............................<a href="#page-8">8</a>
      5.3. Receiving Recording Indications and Providing Recording
           Preferences ................................................<a href="#page-9">9</a>
   <a href="#section-6">6</a>. SIP Handling ...................................................<a href="#page-11">11</a>
      <a href="#section-6.1">6.1</a>. Procedures at the SRC .....................................<a href="#page-11">11</a>
           <a href="#section-6.1.1">6.1.1</a>. Initiating a Recording Session .....................<a href="#page-11">11</a>
           6.1.2. SIP Extensions for Recording Indications
                  and Preferences ....................................<a href="#page-12">12</a>
      <a href="#section-6.2">6.2</a>. Procedures at the SRS .....................................<a href="#page-12">12</a>
      <a href="#section-6.3">6.3</a>. Procedures for Recording-Aware User Agents ................<a href="#page-12">12</a>
   <a href="#section-7">7</a>. SDP Handling ...................................................<a href="#page-13">13</a>
      <a href="#section-7.1">7.1</a>. Procedures at the SRC .....................................<a href="#page-13">13</a>
           <a href="#section-7.1.1">7.1.1</a>. SDP Handling in the RS .............................<a href="#page-13">13</a>
                  <a href="#section-7.1.1.1">7.1.1.1</a>. Handling Media Stream Updates .............<a href="#page-14">14</a>
           <a href="#section-7.1.2">7.1.2</a>. Recording Indication in the CS .....................<a href="#page-15">15</a>
           <a href="#section-7.1.3">7.1.3</a>. Recording Preference in the CS .....................<a href="#page-16">16</a>
      <a href="#section-7.2">7.2</a>. Procedures at the SRS .....................................<a href="#page-16">16</a>
      <a href="#section-7.3">7.3</a>. Procedures for Recording-Aware User Agents ................<a href="#page-18">18</a>
           <a href="#section-7.3.1">7.3.1</a>. Recording Indication ...............................<a href="#page-18">18</a>
           <a href="#section-7.3.2">7.3.2</a>. Recording Preference ...............................<a href="#page-19">19</a>
   <a href="#section-8">8</a>. RTP Handling ...................................................<a href="#page-20">20</a>
      <a href="#section-8.1">8.1</a>. RTP Mechanisms ............................................<a href="#page-20">20</a>
           <a href="#section-8.1.1">8.1.1</a>. RTCP ...............................................<a href="#page-20">20</a>
           <a href="#section-8.1.2">8.1.2</a>. RTP Profile ........................................<a href="#page-21">21</a>
           <a href="#section-8.1.3">8.1.3</a>. SSRC ...............................................<a href="#page-21">21</a>



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           <a href="#section-8.1.4">8.1.4</a>. CSRC ...............................................<a href="#page-22">22</a>
           <a href="#section-8.1.5">8.1.5</a>. SDES ...............................................<a href="#page-22">22</a>
                  <a href="#section-8.1.5.1">8.1.5.1</a>. CNAME .....................................<a href="#page-22">22</a>
           <a href="#section-8.1.6">8.1.6</a>. Keepalive ..........................................<a href="#page-22">22</a>
           <a href="#section-8.1.7">8.1.7</a>. RTCP Feedback Messages .............................<a href="#page-23">23</a>
                  <a href="#section-8.1.7.1">8.1.7.1</a>. Full Intra Request ........................<a href="#page-23">23</a>
                  <a href="#section-8.1.7.2">8.1.7.2</a>. Picture Loss Indication ...................<a href="#page-23">23</a>
                  8.1.7.3. Temporary Maximum Media Stream Bit
                           Rate Request ..............................<a href="#page-24">24</a>
           <a href="#section-8.1.8">8.1.8</a>. Symmetric RTP/RTCP for Sending and Receiving .......<a href="#page-24">24</a>
      <a href="#section-8.2">8.2</a>. Roles .....................................................<a href="#page-25">25</a>
           <a href="#section-8.2.1">8.2.1</a>. SRC Acting as an RTP Translator ....................<a href="#page-26">26</a>
                  <a href="#section-8.2.1.1">8.2.1.1</a>. Forwarding Translator .....................<a href="#page-26">26</a>
                  <a href="#section-8.2.1.2">8.2.1.2</a>. Transcoding Translator ....................<a href="#page-26">26</a>
           <a href="#section-8.2.2">8.2.2</a>. SRC Acting as an RTP Mixer .........................<a href="#page-27">27</a>
           <a href="#section-8.2.3">8.2.3</a>. SRC Acting as an RTP Endpoint ......................<a href="#page-28">28</a>
      <a href="#section-8.3">8.3</a>. RTP Session Usage by SRC ..................................<a href="#page-28">28</a>
           <a href="#section-8.3.1">8.3.1</a>. SRC Using Multiple m-lines .........................<a href="#page-28">28</a>
           <a href="#section-8.3.2">8.3.2</a>. SRC Using Mixing ...................................<a href="#page-29">29</a>
      <a href="#section-8.4">8.4</a>. RTP Session Usage by SRS ..................................<a href="#page-30">30</a>
   <a href="#section-9">9</a>. Metadata .......................................................<a href="#page-31">31</a>
      <a href="#section-9.1">9.1</a>. Procedures at the SRC .....................................<a href="#page-31">31</a>
      <a href="#section-9.2">9.2</a>. Procedures at the SRS .....................................<a href="#page-33">33</a>
   <a href="#section-10">10</a>. Persistent Recording ..........................................<a href="#page-35">35</a>
   <a href="#section-11">11</a>. IANA Considerations ...........................................<a href="#page-36">36</a>
      <a href="#section-11.1">11.1</a>. Registration of Option Tags ..............................<a href="#page-36">36</a>
           <a href="#section-11.1.1">11.1.1</a>. "siprec" Option Tag ...............................<a href="#page-36">36</a>
           <a href="#section-11.1.2">11.1.2</a>. "record-aware" Option Tag .........................<a href="#page-36">36</a>
      <a href="#section-11.2">11.2</a>. Registration of Media Feature Tags .......................<a href="#page-36">36</a>
           <a href="#section-11.2.1">11.2.1</a>. Feature Tag for the SRC ...........................<a href="#page-36">36</a>
           <a href="#section-11.2.2">11.2.2</a>. Feature Tag for the SRS ...........................<a href="#page-37">37</a>
      <a href="#section-11.3">11.3</a>. New Content-Disposition Parameter Registrations ..........<a href="#page-37">37</a>
      <a href="#section-11.4">11.4</a>. SDP Attributes ...........................................<a href="#page-38">38</a>
           <a href="#section-11.4.1">11.4.1</a>. "record" SDP Attribute ............................<a href="#page-38">38</a>
           <a href="#section-11.4.2">11.4.2</a>. "recordpref" SDP Attribute ........................<a href="#page-38">38</a>
   <a href="#section-12">12</a>. Security Considerations .......................................<a href="#page-39">39</a>
      <a href="#section-12.1">12.1</a>. Authentication and Authorization .........................<a href="#page-39">39</a>
      <a href="#section-12.2">12.2</a>. RTP Handling .............................................<a href="#page-40">40</a>
      <a href="#section-12.3">12.3</a>. Metadata .................................................<a href="#page-41">41</a>
      <a href="#section-12.4">12.4</a>. Storage and Playback .....................................<a href="#page-41">41</a>
   <a href="#section-13">13</a>. References ....................................................<a href="#page-41">41</a>
      <a href="#section-13.1">13.1</a>. Normative References .....................................<a href="#page-41">41</a>
      <a href="#section-13.2">13.2</a>. Informative References ...................................<a href="#page-42">42</a>
   Acknowledgements ..................................................<a href="#page-44">44</a>
   Authors' Addresses ................................................<a href="#page-45">45</a>






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

   This document specifies the mechanism to record a Communication
   Session (CS) by delivering real-time media and metadata from the CS
   to a recording device.  In accordance with the architecture
   [<a href="./rfc7245" title="&quot;An Architecture for Media Recording Using the Session Initiation Protocol&quot;">RFC7245</a>], the Session Recording Protocol specifies the use of SIP,
   the Session Description Protocol (SDP), and RTP to establish a
   Recording Session (RS) between the Session Recording Client (SRC),
   which is on the path of the CS, and a Session Recording Server (SRS)
   at the recording device.  SIP is also used to deliver metadata to the
   recording device, as specified in [<a href="./rfc7865" title="&quot;Session Initiation Protocol (SIP) Recording Metadata&quot;">RFC7865</a>].  Metadata is information
   that describes recorded media and the CS to which they relate.  The
   Session Recording Protocol intends to satisfy the SIP-based Media
   Recording (SIPREC) requirements listed in [<a href="./rfc6341" title="&quot;Use Cases and Requirements for SIP-Based Media Recording (SIPREC)&quot;">RFC6341</a>].  In addition to
   the Session Recording Protocol, this document specifies extensions
   for user agents (UAs) that are participants in a CS to receive
   recording indications and to provide preferences for recording.

   This document considers only active recording, where the SRC
   purposefully streams media to an SRS and all participating UAs are
   notified of the recording.  Passive recording, where a recording
   device detects media directly from the network (e.g., using
   port-mirroring techniques), is outside the scope of this document.
   In addition, lawful intercept is outside the scope of this document,
   in accordance with [<a href="./rfc2804" title="&quot;IETF Policy on Wiretapping&quot;">RFC2804</a>].

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

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in [<a href="./rfc2119" title="&quot;Key words for use in RFCs to Indicate Requirement Levels&quot;">RFC2119</a>].

<span class="h2"><a class="selflink" id="section-3" href="#section-3">3</a>.  Definitions</span>

   This document refers to the core definitions provided in the
   architecture document [<a href="./rfc7245" title="&quot;An Architecture for Media Recording Using the Session Initiation Protocol&quot;">RFC7245</a>].

   <a href="#section-8">Section 8</a> uses the definitions provided in "RTP: A Transport Protocol
   for Real-Time Applications" [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>].

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

   The scope of the Session Recording Protocol includes the
   establishment of the RSs and the reporting of the metadata.  The
   scope also includes extensions supported by UAs participating in the
   CS, such as an indication of recording.  The UAs need not be
   recording aware in order to participate in a CS being recorded.




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   The items in the following list, which is not exhaustive, do not
   represent the protocol itself and are considered out of scope for the
   Session Recording Protocol:

   o  Delivering recorded media in real time as the CS media

   o  Specifications of criteria to select a specific CS to be recorded
      or triggers to record a certain CS in the future

   o  Recording policies that determine whether the CS should be
      recorded and whether parts of the CS are to be recorded

   o  Retention policies that determine how long a recording is stored

   o  Searching and accessing the recorded media and metadata

   o  Policies governing how CS users are made aware of recording

   o  Delivering additional RS metadata through a non-SIP mechanism

<span class="h2"><a class="selflink" id="section-5" href="#section-5">5</a>.  Overview of Operations</span>

   This section is informative and provides a description of recording
   operations.

   <a href="#section-6">Section 6</a> describes the SIP communication in an RS between an SRC and
   an SRS, as well as the procedures for recording-aware UAs
   participating in a CS.  <a href="#section-7">Section 7</a> describes SDP handling in an RS,
   and the procedures for recording indications and recording
   preferences.  <a href="#section-8">Section 8</a> describes RTP handling in an RS.  <a href="#section-9">Section 9</a>
   describes the mechanism to deliver recording metadata from the SRC to
   the SRS.

   As mentioned in the architecture document [<a href="./rfc7245" title="&quot;An Architecture for Media Recording Using the Session Initiation Protocol&quot;">RFC7245</a>], there are a
   number of types of call flows based on the location of the SRC.  The
   sample call flows discussed in <a href="#section-5.1">Section 5.1</a> provide a quick overview
   of the operations between the SRC and the SRS.

<span class="h3"><a class="selflink" id="section-5.1" href="#section-5.1">5.1</a>.  Delivering Recorded Media</span>

   When a SIP Back-to-Back User Agent (B2BUA) with SRC functionality
   routes a call from UA A to UA B, the SRC has access to the media path
   between the UAs.  When the SRC is aware that it should be recording
   the conversation, the SRC can cause the B2BUA to relay the media
   between UA A and UA B.  The SRC then establishes the RS with the SRS
   and sends replicated media towards the SRS.





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   An endpoint may also have SRC functionality, where the endpoint
   itself establishes the RS to the SRS.  Since the endpoint has access
   to the media in the CS, the endpoint can send replicated media
   towards the SRS.

   The example call flows in Figures 1 and 2 show an SRC establishing an
   RS towards an SRS.  Figure 1 illustrates UA A acting as the SRC.
   Figure 2 illustrates a B2BUA acting as the SRC.  Note that the SRC
   can choose when to establish the RS independent of the CS, even
   though the example call flows suggest that the SRC is establishing
   the RS (message (5) in Figure 2) after the CS is established.

            UA A/SRC               UA B                    SRS
             |(1) CS INVITE          |                      |
             |----------------------&gt;|                      |
             |           (2) 200 OK  |                      |
             |&lt;----------------------|                      |
             |                       |                      |
             |(3) RS INVITE with SDP |                      |
             |---------------------------------------------&gt;|
             |                       |  (4) 200 OK with SDP |
             |&lt;---------------------------------------------|
             |(5) CS RTP             |                      |
             |======================&gt;|                      |
             |&lt;======================|                      |
             |(6) RS RTP             |                      |
             |=============================================&gt;|
             |=============================================&gt;|
             |                       |                      |
             |(7) CS BYE             |                      |
             |----------------------&gt;|                      |
             |(8) RS BYE             |                      |
             |---------------------------------------------&gt;|
             |                       |                      |

            Figure 1: Basic Recording Call Flow with UA as SRC















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     UA A           SRC                    UA B                    SRS
      |(1) CS INVITE |                       |                      |
      |-------------&gt;|                       |                      |
      |              |(2) CS INVITE          |                      |
      |              |----------------------&gt;|                      |
      |              |           (3) 200 OK  |                      |
      |              |&lt;----------------------|                      |
      |   (4) 200 OK |                       |                      |
      |&lt;-------------|                       |                      |
      |              |(5) RS INVITE with SDP |                      |
      |              |---------------------------------------------&gt;|
      |              |                       |  (6) 200 OK with SDP |
      |              |&lt;---------------------------------------------|
      |(7) CS RTP    |                       |                      |
      |=============&gt;|======================&gt;|                      |
      |&lt;=============|&lt;======================|                      |
      |              |(8) RS RTP             |                      |
      |              |=============================================&gt;|
      |              |=============================================&gt;|
      |(9) CS BYE    |                       |                      |
      |-------------&gt;|                       |                      |
      |              |(10) CS BYE            |                      |
      |              |----------------------&gt;|                      |
      |              |(11) RS BYE            |                      |
      |              |---------------------------------------------&gt;|
      |              |                       |                      |

           Figure 2: Basic Recording Call Flow with B2BUA as SRC

   The call flow shown in Figure 2 can also apply to the case of a
   centralized conference with a mixer.  For clarity, ACKs to INVITEs
   and 200 OKs to BYEs are not shown.  The conference focus can provide
   the SRC functionality, since the conference focus has access to all
   the media from each conference participant.  When a recording is
   requested, the SRC delivers the metadata and the media streams to the
   SRS.  Since the conference focus has access to a mixer, the SRC may
   choose to mix the media streams from all participants as a single
   mixed media stream towards the SRS.

   An SRC can use a single RS to record multiple CSs.  Every time the
   SRC wants to record a new call, the SRC updates the RS with a new SDP
   offer to add new recorded streams to the RS and to correspondingly
   also update the metadata for the new call.

   An SRS can also establish an RS to an SRC, although it is beyond the
   scope of this document to define how an SRS would specify which calls
   to record.




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<span class="h3"><a class="selflink" id="section-5.2" href="#section-5.2">5.2</a>.  Delivering Recording Metadata</span>

   The SRC is responsible for the delivery of metadata to the SRS.  The
   SRC may provide an initial metadata snapshot about recorded media
   streams in the initial INVITE content in the RS.  Subsequent metadata
   updates can be represented as a stream of events in UPDATE [<a href="./rfc3311" title="&quot;The Session Initiation Protocol (SIP) UPDATE Method&quot;">RFC3311</a>]
   or re-INVITE requests sent by the SRC.  These metadata updates are
   normally incremental updates to the initial metadata snapshot to
   optimize on the size of updates.  However, the SRC may also decide to
   send a new metadata snapshot at any time.

   Metadata is transported in the body of INVITE or UPDATE messages.
   Certain metadata, such as the attributes of the recorded media
   stream, is located in the SDP of the RS.

   The SRS has the ability to send a request to the SRC to ask for a new
   metadata snapshot update from the SRC.  This can happen when the SRS
   fails to understand the current stream of incremental updates for
   whatever reason -- for example, when the SRS loses the current state
   due to internal failure.  The SRS may optionally attach a reason
   along with the snapshot request.  This request allows both the SRC
   and the SRS to synchronize the states with a new metadata snapshot so
   that further incremental metadata updates will be based on the latest
   metadata snapshot.  Similar to the metadata content, the metadata
   snapshot request is transported as content in UPDATE or INVITE
   messages sent by the SRS in the RS.

























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          SRC                                                   SRS
           |                                                     |
           |(1) INVITE (metadata snapshot 1)                     |
           |----------------------------------------------------&gt;|
           |                                          (2) 200 OK |
           |&lt;----------------------------------------------------|
           |(3) ACK                                              |
           |----------------------------------------------------&gt;|
           |(4) RTP                                              |
           |====================================================&gt;|
           |====================================================&gt;|
           |(5) UPDATE (metadata update 1)                       |
           |----------------------------------------------------&gt;|
           |                                          (6) 200 OK |
           |&lt;----------------------------------------------------|
           |(7) UPDATE (metadata update 2)                       |
           |----------------------------------------------------&gt;|
           |                                          (8) 200 OK |
           |&lt;----------------------------------------------------|
           |              (9) UPDATE (metadata snapshot request) |
           |&lt;----------------------------------------------------|
           |                                        (10) 200 OK  |
           |----------------------------------------------------&gt;|
           |      (11) INVITE (metadata snapshot 2 + SDP offer)  |
           |----------------------------------------------------&gt;|
           |                            (12) 200 OK (SDP answer) |
           |&lt;----------------------------------------------------|
           | (13) UPDATE (metadata update 1 based on snapshot 2) |
           |----------------------------------------------------&gt;|
           |                                         (14) 200 OK |
           |&lt;----------------------------------------------------|

               Figure 3: Delivering Metadata via SIP UPDATE

<span class="h3"><a class="selflink" id="section-5.3" href="#section-5.3">5.3</a>.  Receiving Recording Indications and Providing Recording</span>
<span class="h3">      Preferences</span>

   The SRC is responsible for providing recording indications to the
   participants in the CS.  A recording-aware UA supports receiving
   recording indications via the SDP "a=record" attribute, and it can
   specify a recording preference in the CS by including the SDP
   "a=recordpref" attribute.  The recording attribute is a declaration
   by the SRC in the CS to indicate whether recording is taking place.
   The recording preference attribute is a declaration by the recording-
   aware UA in the CS to indicate its recording preference.  A UA that
   does not want to be recorded may still be notified that recording is
   occurring, for a number of reasons (e.g., it was not capable of




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   indicating its preference, its preference was ignored).  If this
   occurs, the UA's only mechanism to avoid being recorded is to
   terminate its participation in the session.

   To illustrate how the attributes are used, if UA A is initiating a
   call to UA B and UA A is also an SRC that is performing the
   recording, then UA A provides the recording indication in the SDP
   offer with a=record:on.  Since UA A is the SRC, UA A receives the
   recording indication from the SRC directly.  When UA B receives the
   SDP offer, UA B will see that recording is happening on the other
   endpoint of this session.  Since UA B is not an SRC and does not
   provide any recording preference, the SDP answer does not contain
   a=record or a=recordpref.

        UA A                                                   UA B
        (SRC)                                                   |
          |                                                     |
          |                [SRC recording starts]               |
          |(1) INVITE (SDP offer + a=record:on)                 |
          |----------------------------------------------------&gt;|
          |                             (2) 200 OK (SDP answer) |
          |&lt;----------------------------------------------------|
          |(3) ACK                                              |
          |----------------------------------------------------&gt;|
          |(4) RTP                                              |
          |&lt;===================================================&gt;|
          |                                                     |
          |   [UA B wants to set preference to no recording]    |
          |           (5) INVITE (SDP offer + a=recordpref:off) |
          |&lt;----------------------------------------------------|
          |   [SRC honors the preference and stops recording]   |
          |(6) 200 OK (SDP answer + a=record:off)               |
          |----------------------------------------------------&gt;|
          |                                             (7) ACK |
          |&lt;----------------------------------------------------|

          Figure 4: Recording Indication and Recording Preference

   After the call is established and recording is in progress, UA B
   later decides to change the recording preference to no recording and
   sends a re-INVITE with the "a=recordpref" attribute.  It is up to the
   SRC to honor the preference, and in this case the SRC decides to stop
   the recording and updates the recording indication in the SDP answer.








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   Note that UA B could have explicitly indicated a recording preference
   in (2), the 200 OK for the original INVITE.  Indicating a preference
   of no recording in an initial INVITE or an initial response to an
   INVITE may reduce the chance of a user being recorded in the
   first place.

<span class="h2"><a class="selflink" id="section-6" href="#section-6">6</a>.  SIP Handling</span>

<span class="h3"><a class="selflink" id="section-6.1" href="#section-6.1">6.1</a>.  Procedures at the SRC</span>

<span class="h4"><a class="selflink" id="section-6.1.1" href="#section-6.1.1">6.1.1</a>.  Initiating a Recording Session</span>

   An RS is a SIP session with specific extensions applied, and these
   extensions are listed in the procedures below for the SRC and the
   SRS.  When an SRC or an SRS receives a SIP session that is not an RS,
   it is up to the SRC or the SRS to determine what to do with the SIP
   session.

   The SRC can initiate an RS by sending a SIP INVITE request to the
   SRS.  The SRC and the SRS are identified in the From and To headers,
   respectively.

   The SRC MUST include the "+sip.src" feature tag in the Contact URI,
   defined in this specification as an extension to [<a href="./rfc3840" title="&quot;Indicating User Agent Capabilities in the Session Initiation Protocol (SIP)&quot;">RFC3840</a>], for all
   RSs.  An SRS uses the presence of the "+sip.src" feature tag in
   dialog creating and modifying requests and responses to confirm that
   the dialog being created is for the purpose of an RS.  In addition,
   when an SRC sends a REGISTER request to a registrar, the SRC MAY
   include the "+sip.src" feature tag to indicate that it is an SRC.

   Since SIP Caller Preferences extensions are optional to implement for
   routing proxies, there is no guarantee that an RS will be routed to
   an SRC or SRS.  A new option tag, "siprec", is introduced.  As per
   [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>], only an SRC or an SRS can accept this option tag in an RS.
   An SRC MUST include the "siprec" option tag in the Require header
   when initiating an RS so that UAs that do not support the Session
   Recording Protocol extensions will simply reject the INVITE request
   with a 420 (Bad Extension) response.

   When an SRC receives a new INVITE, the SRC MUST only consider the SIP
   session as an RS when both the "+sip.srs" feature tag and the
   "siprec" option tag are included in the INVITE request.









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<span class="h4"><a class="selflink" id="section-6.1.2" href="#section-6.1.2">6.1.2</a>.  SIP Extensions for Recording Indications and Preferences</span>

   For the CS, the SRC MUST provide recording indications to all
   participants in the CS.  A participant UA in a CS can indicate that
   it is recording aware by providing the "record-aware" option tag, and
   the SRC MUST provide recording indications in the new SDP "a=record"
   attribute described in <a href="#section-7">Section 7</a> below.  In the absence of the
   "record-aware" option tag -- meaning that the participant UA is not
   recording aware -- an SRC MUST provide recording indications through
   other means, such as playing a tone in-band or having a signed
   participant contract in place.

   An SRC in the CS may also indicate itself as a session recording
   client by including the "+sip.src" feature tag.  A recording-aware
   participant can learn that an SRC is in the CS and can set the
   recording preference for the CS with the new SDP "a=recordpref"
   attribute described in <a href="#section-7">Section 7</a>.

<span class="h3"><a class="selflink" id="section-6.2" href="#section-6.2">6.2</a>.  Procedures at the SRS</span>

   When an SRS receives a new INVITE, the SRS MUST only consider the SIP
   session as an RS when both the "+sip.src" feature tag and the
   "siprec" option tag are included in the INVITE request.

   The SRS can initiate an RS by sending a SIP INVITE request to the
   SRC.  The SRS and the SRC are identified in the From and To headers,
   respectively.

   The SRS MUST include the "+sip.srs" feature tag in the Contact URI,
   as per [<a href="./rfc3840" title="&quot;Indicating User Agent Capabilities in the Session Initiation Protocol (SIP)&quot;">RFC3840</a>], for all RSs.  An SRC uses the presence of this
   feature tag in dialog creation and modification requests and
   responses to confirm that the dialog being created is for the purpose
   of an RS (REQ-030 in [<a href="./rfc6341" title="&quot;Use Cases and Requirements for SIP-Based Media Recording (SIPREC)&quot;">RFC6341</a>]).  In addition, when an SRS sends a
   REGISTER request to a registrar, the SRS SHOULD include the
   "+sip.srs" feature tag to indicate that it is an SRS.

   An SRS MUST include the "siprec" option tag in the Require header as
   per [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>] when initiating an RS so that UAs that do not support
   the Session Recording Protocol extensions will simply reject the
   INVITE request with a 420 (Bad Extension) response.

<span class="h3"><a class="selflink" id="section-6.3" href="#section-6.3">6.3</a>.  Procedures for Recording-Aware User Agents</span>

   A recording-aware UA is a participant in the CS that supports the SIP
   and SDP extensions for receiving recording indications and for
   requesting recording preferences for the call.  A recording-aware UA
   MUST indicate that it can accept the reporting of recording
   indications provided by the SRC with a new "record-aware" option tag



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   when initiating or establishing a CS; this means including the
   "record-aware" option tag in the Supported header in the initial
   INVITE request or response.

   A recording-aware UA MUST provide a recording indication to the end
   user through an appropriate user interface, indicating whether
   recording is on, off, or paused for each medium.  Appropriate user
   interfaces may include real-time notification or previously
   established agreements that use of the device is subject to
   recording.  Some UAs that are automatons (e.g., Interactive Voice
   Response (IVR), media server, Public Switched Telephone Network
   (PSTN) gateway) may not have a user interface to render a recording
   indication.  When such a UA indicates recording awareness, the UA
   SHOULD render the recording indication through other means, such as
   passing an in-band tone on the PSTN gateway, putting the recording
   indication in a log file, or raising an application event in a
   VoiceXML dialog.  These UAs MAY also choose not to indicate recording
   awareness, thereby relying on whatever mechanism an SRC chooses to
   indicate recording, such as playing a tone in-band.

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

<span class="h3"><a class="selflink" id="section-7.1" href="#section-7.1">7.1</a>.  Procedures at the SRC</span>

   The SRC and SRS follow the SDP offer/answer model described in
   [<a href="./rfc3264" title="&quot;An Offer/Answer Model with Session Description Protocol (SDP)&quot;">RFC3264</a>].  The procedures for the SRC and SRS describe the
   conventions used in an RS.

<span class="h4"><a class="selflink" id="section-7.1.1" href="#section-7.1.1">7.1.1</a>.  SDP Handling in the RS</span>

   Since the SRC does not expect to receive media from the SRS, the SRC
   typically sets each media stream of the SDP offer to only send media,
   by qualifying them with the "a=sendonly" attribute, according to the
   procedures in [<a href="./rfc3264" title="&quot;An Offer/Answer Model with Session Description Protocol (SDP)&quot;">RFC3264</a>].

   The SRC sends recorded streams of participants to the SRS, and the
   SRC MUST provide a "label" attribute ("a=label"), as per [<a href="./rfc4574" title="&quot;The Session Description Protocol (SDP) Label Attribute&quot;">RFC4574</a>],
   on each media stream in order to identify the recorded stream with
   the rest of the metadata.  The "a=label" attribute identifies each
   recorded media stream, and the label name is mapped to the Media
   Stream Reference in the metadata as per [<a href="./rfc7865" title="&quot;Session Initiation Protocol (SIP) Recording Metadata&quot;">RFC7865</a>].  The scope of the
   "a=label" attribute only applies to the SDP and metadata conveyed in
   the bodies of the SIP request or response that the label appeared in.
   Note that a recorded stream is distinct from a CS stream; the
   metadata provides a list of participants that contribute to each
   recorded stream.





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   Figure 5 shows an example SDP offer from an SRC with both audio and
   video recorded streams.  Note that this example contains unfolded
   lines longer than 72 characters; these lines are captured between
   &lt;allOneLine&gt; tags.

       v=0
       o=SRC 2890844526 2890844526 IN IP4 198.51.100.1
       s=-
       c=IN IP4 198.51.100.1
       t=0 0
       m=audio 12240 RTP/AVP 0 4 8
       a=sendonly
       a=label:1
       m=video 22456 RTP/AVP 98
       a=rtpmap:98 H264/90000
       &lt;allOneLine&gt;
       a=fmtp:98 profile-level-id=42A01E;
                 sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==
       &lt;/allOneLine&gt;
       a=sendonly
       a=label:2
       m=audio 12242 RTP/AVP 0 4 8
       a=sendonly
       a=label:3
       m=video 22458 RTP/AVP 98
       a=rtpmap:98 H264/90000
       &lt;allOneLine&gt;
       a=fmtp:98 profile-level-id=42A01E;
                 sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==
       &lt;/allOneLine&gt;
       a=sendonly
       a=label:4

     Figure 5: Sample SDP Offer from SRC with Audio and Video Streams

<span class="h5"><a class="selflink" id="section-7.1.1.1" href="#section-7.1.1.1">7.1.1.1</a>.  Handling Media Stream Updates</span>

   Over the lifetime of an RS, the SRC can add and remove recorded
   streams to and from the RS for various reasons -- for example, when a
   CS stream is added to or removed from the CS, or when a CS is created
   or terminated if an RS handles multiple CSs.  To remove a recorded
   stream from the RS, the SRC sends a new SDP offer where the port of
   the media stream to be removed is set to zero, according to the
   procedures in [<a href="./rfc3264" title="&quot;An Offer/Answer Model with Session Description Protocol (SDP)&quot;">RFC3264</a>].  To add a recorded stream to the RS, the SRC
   sends a new SDP offer by adding a new media stream description or by
   reusing an old media stream that had been previously disabled,
   according to the procedures in [<a href="./rfc3264" title="&quot;An Offer/Answer Model with Session Description Protocol (SDP)&quot;">RFC3264</a>].




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   The SRC can temporarily discontinue streaming and collection of
   recorded media from the SRC to the SRS for reasons such as masking
   the recording.  In this case, the SRC sends a new SDP offer and sets
   the media stream to inactive (a=inactive) for each recorded stream to
   be paused, as per the procedures in [<a href="./rfc3264" title="&quot;An Offer/Answer Model with Session Description Protocol (SDP)&quot;">RFC3264</a>].  To resume streaming
   and collection of recorded media, the SRC sends a new SDP offer and
   sets the media stream to sendonly (a=sendonly).  Note that a CS may
   itself change the media stream direction by updating the SDP -- for
   example, by setting a=inactive for SDP hold.  Media stream direction
   changes in the CS are conveyed in the metadata by the SRC.  When a CS
   media stream is changed to or from inactive, the effect on the
   corresponding RS media stream is governed by SRC policy.  The SRC MAY
   have a local policy to pause an RS media stream when the
   corresponding CS media stream is inactive, or it MAY leave the RS
   media stream as sendonly.

<span class="h4"><a class="selflink" id="section-7.1.2" href="#section-7.1.2">7.1.2</a>.  Recording Indication in the CS</span>

   While there are existing mechanisms for providing an indication that
   a CS is being recorded, these mechanisms are usually delivered on the
   CS media streams, such as playing an in-band tone or an announcement
   to the participants.  A new "record" SDP attribute is introduced to
   allow the SRC to indicate recording state to a recording-aware UA in
   a CS.

   The "record" SDP attribute appears at the media level or
   session level in either an SDP offer or answer.  When the attribute
   is applied at the session level, the indication applies to all media
   streams in the SDP.  When the attribute is applied at the
   media level, the indication applies to that one media stream only,
   and that overrides the indication if also set at the session level.
   Whenever the recording indication needs to change, such as
   termination of recording, the SRC MUST initiate a re-INVITE or UPDATE
   to update the SDP "a=record" attribute.

   The following is the ABNF [<a href="./rfc5234" title="&quot;Augmented BNF for Syntax Specifications: ABNF&quot;">RFC5234</a>] of the "record" attribute:

       attribute =/ record-attr
       ; attribute defined in <a href="./rfc4566">RFC 4566</a>

       record-attr = "record:" indication
       indication = "on" / "off" / "paused"

   on:      Recording is in progress.

   off:     No recording is in progress.

   paused:  Recording is in progress but media is paused.



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<span class="h4"><a class="selflink" id="section-7.1.3" href="#section-7.1.3">7.1.3</a>.  Recording Preference in the CS</span>

   When the SRC receives the "a=recordpref" SDP in an SDP offer or
   answer, the SRC chooses to honor the preference to record based on
   local policy at the SRC.  If the SRC makes a change in recording
   state, the SRC MUST report the new recording state in the "a=record"
   attribute in the SDP answer or in a subsequent SDP offer.

<span class="h3"><a class="selflink" id="section-7.2" href="#section-7.2">7.2</a>.  Procedures at the SRS</span>

   Typically, the SRS only receives RTP streams from the SRC; therefore,
   the SDP offer/answer from the SRS normally sets each media stream to
   receive media, by setting them with the "a=recvonly" attribute,
   according to the procedures of [<a href="./rfc3264" title="&quot;An Offer/Answer Model with Session Description Protocol (SDP)&quot;">RFC3264</a>].  When the SRS is not ready
   to receive a recorded stream, the SRS sets the media stream as
   inactive in the SDP offer or answer by setting it with an
   "a=inactive" attribute, according to the procedures of [<a href="./rfc3264" title="&quot;An Offer/Answer Model with Session Description Protocol (SDP)&quot;">RFC3264</a>].
   When the SRS is ready to receive recorded streams, the SRS sends a
   new SDP offer and sets the media streams with an "a=recvonly"
   attribute.































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   Figure 6 shows an example of an SDP answer from the SRS for the SDP
   offer from Figure 5.  Note that this example contains unfolded lines
   longer than 72 characters; these lines are captured between
   &lt;allOneLine&gt; tags.

       v=0
       o=SRS 0 0 IN IP4 198.51.100.20
       s=-
       c=IN IP4 198.51.100.20
       t=0 0
       m=audio 10000 RTP/AVP 0
       a=recvonly
       a=label:1
       m=video 10002 RTP/AVP 98
       a=rtpmap:98 H264/90000
       &lt;allOneLine&gt;
       a=fmtp:98 profile-level-id=42A01E;
                 sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==
       &lt;/allOneLine&gt;
       a=recvonly
       a=label:2
       m=audio 10004 RTP/AVP 0
       a=recvonly
       a=label:3
       m=video 10006 RTP/AVP 98
       a=rtpmap:98 H264/90000
       &lt;allOneLine&gt;
       a=fmtp:98 profile-level-id=42A01E;
                 sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==
       &lt;/allOneLine&gt;
       a=recvonly
       a=label:4

     Figure 6: Sample SDP Answer from SRS with Audio and Video Streams

   Over the lifetime of an RS, the SRS can remove recorded streams from
   the RS for various reasons.  To remove a recorded stream from the RS,
   the SRS sends a new SDP offer where the port of the media stream to
   be removed is set to zero, according to the procedures in [<a href="./rfc3264" title="&quot;An Offer/Answer Model with Session Description Protocol (SDP)&quot;">RFC3264</a>].

   The SRS MUST NOT add recorded streams in the RS when the SRS sends a
   new SDP offer.  Similarly, when the SRS starts an RS, the SRS MUST
   initiate the INVITE without an SDP offer to let the SRC generate the
   SDP offer with the streams to be recorded.







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   The sequence diagram in Figure 7 shows an example where the SRS is
   initially not ready to receive recorded streams and later updates the
   RS when the SRS is ready to record.

     SRC                                                   SRS
      |                                                     |
      |(1) INVITE (SDP offer)                               |
      |----------------------------------------------------&gt;|
      |                                           [not ready to record]
      |                        (2) 200 OK with SDP inactive |
      |&lt;----------------------------------------------------|
      |(3) ACK                                              |
      |----------------------------------------------------&gt;|
      |                      ...                            |
      |                                             [ready to record]
      |                     (4) re-INVITE with SDP recvonly |
      |&lt;----------------------------------------------------|
      |(5) 200 OK with SDP sendonly                         |
      |----------------------------------------------------&gt;|
      |                                             (6) ACK |
      |&lt;----------------------------------------------------|
      |(7) RTP                                              |
      |====================================================&gt;|
      |                      ...                            |
      |(8) BYE                                              |
      |----------------------------------------------------&gt;|
      |                                             (9) OK  |
      |&lt;----------------------------------------------------|

             Figure 7: SRS Responding to Offer with a=inactive

<span class="h3"><a class="selflink" id="section-7.3" href="#section-7.3">7.3</a>.  Procedures for Recording-Aware User Agents</span>

<span class="h4"><a class="selflink" id="section-7.3.1" href="#section-7.3.1">7.3.1</a>.  Recording Indication</span>

   When a recording-aware UA receives an SDP offer or answer that
   includes the "a=record" attribute, the UA provides to the end user an
   indication as to whether the recording is on, off, or paused for each
   medium, based on the most recently received "a=record" SDP attribute
   for that medium.

   When a CS is traversed through multiple UAs such as a B2BUA or a
   conference focus, each UA involved in the CS that is aware that the
   CS is being recorded MUST provide the recording indication through
   the "a=record" attribute to all other parties in the CS.






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   It is possible that more than one SRC is in the call path of the same
   CS, but the recording indication attribute does not provide any hint
   as to which SRC or how many SRCs are recording.  An endpoint knows
   only that the call is being recorded.  Furthermore, this attribute is
   not used as a request for a specific SRC to start or stop recording.

<span class="h4"><a class="selflink" id="section-7.3.2" href="#section-7.3.2">7.3.2</a>.  Recording Preference</span>

   A participant in a CS MAY set the recording preference in the CS to
   be recorded or not recorded at session establishment or during the
   session.  A new "recordpref" SDP attribute is introduced, and the
   participant in the CS may set this recording preference attribute in
   any SDP offer/answer at session establishment time or during the
   session.  The SRC is not required to honor the recording preference
   from a participant, based on local policies at the SRC, and the
   participant can learn the recording indication through the "a=record"
   SDP attribute as described in <a href="#section-7.3.1">Section 7.3.1</a>.

   The SDP "a=recordpref" attribute can appear at the media level or
   session level and can appear in an SDP offer or answer.  When the
   attribute is applied at the session level, the recording preference
   applies to all media streams in the SDP.  When the attribute is
   applied at the media level, the recording preference applies to that
   one media stream only, and that overrides the recording preference if
   also set at the session level.  The UA can change the recording
   preference by changing the "a=recordpref" attribute in a subsequent
   SDP offer or answer.  The absence of the "a=recordpref" attribute in
   the SDP indicates that the UA has no recording preference.

   The following is the ABNF of the "recordpref" attribute:

       attribute =/ recordpref-attr
       ; attribute defined in <a href="./rfc4566">RFC 4566</a>

       recordpref-attr = "a=recordpref:" pref
       pref = "on" / "off" / "pause" / "nopreference"

   on:     Sets the preference to record if it has not already been
           started.  If the recording is currently paused, the
           preference is to resume recording.

   off:    Sets the preference for no recording.  If recording has
           already been started, then the preference is to stop the
           recording.







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   pause:  If the recording is currently in progress, sets the
           preference to pause the recording.

   nopreference:
           Indicates that the UA has no preference regarding recording.

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

   This section provides recommendations and guidelines for RTP and the
   Real-time Transport Control Protocol (RTCP) in the context of SIPREC
   [<a href="./rfc6341" title="&quot;Use Cases and Requirements for SIP-Based Media Recording (SIPREC)&quot;">RFC6341</a>].  In order to communicate most effectively, the SRC, the
   SRS, and any recording-aware UAs should utilize the mechanisms
   provided by RTP in a well-defined and predictable manner.  It is the
   goal of this document to make the reader aware of these mechanisms
   and to provide recommendations and guidelines.

<span class="h3"><a class="selflink" id="section-8.1" href="#section-8.1">8.1</a>.  RTP Mechanisms</span>

   This section briefly describes important RTP/RTCP constructs and
   mechanisms that are particularly useful within the context of SIPREC.

<span class="h4"><a class="selflink" id="section-8.1.1" href="#section-8.1.1">8.1.1</a>.  RTCP</span>

   The RTP data transport is augmented by a control protocol (RTCP) to
   allow monitoring of the data delivery.  RTCP, as defined in
   [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>], is based on the periodic transmission of control packets
   to all participants in the RTP session, using the same distribution
   mechanism as the data packets.  Support for RTCP is REQUIRED, per
   [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>], and it provides, among other things, the following
   important functionality in relation to SIPREC:

   1) Feedback on the quality of the data distribution

      This feedback from the receivers may be used to diagnose faults in
      the distribution.  As such, RTCP is a well-defined and efficient
      mechanism for the SRS to inform the SRC, and for the SRC to inform
      recording-aware UAs, of issues that arise with respect to the
      reception of media that is to be recorded.

   2) Including a persistent transport-level identifier -- the CNAME, or
      canonical name -- for an RTP source

      The synchronization source (SSRC) [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>] identifier may change
      if a conflict is discovered or a program is restarted, in which
      case receivers can use the CNAME to keep track of each
      participant.  Receivers may also use the CNAME to associate





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      multiple data streams from a given participant in a set of related
      RTP sessions -- for example, to synchronize audio and video.
      Synchronization of media streams is also facilitated by the NTP
      and RTP timestamps included in RTCP packets by data senders.

<span class="h4"><a class="selflink" id="section-8.1.2" href="#section-8.1.2">8.1.2</a>.  RTP Profile</span>

   The RECOMMENDED RTP profiles for the SRC, SRS, and recording-aware
   UAs are "Extended Secure RTP Profile for Real-time Transport Control
   Protocol (RTCP)-Based Feedback (RTP/SAVPF)" [<a href="./rfc5124" title="&quot;Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)&quot;">RFC5124</a>] when using
   encrypted RTP streams, and "Extended RTP Profile for Real-time
   Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)"
   [<a href="./rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>] when using non-encrypted media streams.  However, as these
   are not requirements, some implementations may use "The Secure
   Real-time Transport Protocol (SRTP)" [<a href="./rfc3711" title="&quot;The Secure Real-time Transport Protocol (SRTP)&quot;">RFC3711</a>] and "RTP Profile for
   Audio and Video Conferences with Minimal Control" [<a href="./rfc3551" title="&quot;RTP Profile for Audio and Video Conferences with Minimal Control&quot;">RFC3551</a>].
   Therefore, it is RECOMMENDED that the SRC, SRS, and recording-aware
   UAs not rely entirely on RTP/SAVPF or RTP/AVPF for core functionality
   that may be at least partially achievable using RTP/SAVP and RTP/AVP.

   AVPF and SAVPF provide an improved RTCP timer model that allows more
   flexible transmission of RTCP packets in response to events, rather
   than strictly according to bandwidth.  AVPF-based codec control
   messages provide efficient mechanisms for an SRC, an SRS, and
   recording-aware UAs to handle events such as scene changes, error
   recovery, and dynamic bandwidth adjustments.  These messages are
   discussed in more detail later in this document.

   SAVP and SAVPF provide media encryption, integrity protection, replay
   protection, and a limited form of source authentication.  They do not
   contain or require a specific keying mechanism.

<span class="h4"><a class="selflink" id="section-8.1.3" href="#section-8.1.3">8.1.3</a>.  SSRC</span>

   The SSRC, as defined in [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>], is carried in the RTP header and
   in various fields of RTCP packets.  It is a random 32-bit number that
   is required to be globally unique within an RTP session.  It is
   crucial that the number be chosen with care, in order that
   participants on the same network or starting at the same time are not
   likely to choose the same number.  Guidelines regarding SSRC value
   selection and conflict resolution are provided in [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>].

   The SSRC may also be used to separate different sources of media
   within a single RTP session.  For this reason, as well as for
   conflict resolution, it is important that the SRC, SRS, and
   recording-aware UAs handle changes in SSRC values and properly
   identify the reason for the change.  The CNAME values carried in RTCP
   facilitate this identification.



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<span class="h4"><a class="selflink" id="section-8.1.4" href="#section-8.1.4">8.1.4</a>.  CSRC</span>

   The contributing source (CSRC), as defined in [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>], identifies
   the source of a stream of RTP packets that has contributed to the
   combined stream produced by an RTP mixer.  The mixer inserts a list
   of the SSRC identifiers of the sources that contributed to the
   generation of a particular packet into the RTP header of that packet.
   This list is called the CSRC list.  It is RECOMMENDED that an SRC or
   recording-aware UA, when acting as a mixer, set the CSRC list
   accordingly, and that the SRC and SRS interpret the CSRC list per
   [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>] when received.

<span class="h4"><a class="selflink" id="section-8.1.5" href="#section-8.1.5">8.1.5</a>.  SDES</span>

   The Source Description (SDES), as defined in [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>], contains an
   SSRC/CSRC identifier followed by a list of zero or more items that
   carry information about the SSRC/CSRC.  End systems send one SDES
   packet containing their own source identifier (the same as the SSRC
   in the fixed RTP header).  A mixer sends one SDES packet containing a
   chunk for each CSRC from which it is receiving SDES information, or
   multiple complete SDES packets if there are more than 31 such
   sources.

   The ability to identify individual CSRCs is important in the context
   of SIPREC.  Metadata [<a href="./rfc7865" title="&quot;Session Initiation Protocol (SIP) Recording Metadata&quot;">RFC7865</a>] provides a mechanism to achieve this
   at the signaling level.  SDES provides a mechanism at the RTP level.

<span class="h5"><a class="selflink" id="section-8.1.5.1" href="#section-8.1.5.1">8.1.5.1</a>.  CNAME</span>

   The Canonical End-Point Identifier (CNAME), as defined in [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>],
   provides the binding from the SSRC identifier to an identifier for
   the source (sender or receiver) that remains constant.  It is
   important that the SRC and recording-aware UAs generate CNAMEs
   appropriately and that the SRC and SRS interpret and use them for
   this purpose.  Guidelines for generating CNAME values are provided in
   "Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names
   (CNAMEs)" [<a href="./rfc7022" title="&quot;Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs)&quot;">RFC7022</a>].

<span class="h4"><a class="selflink" id="section-8.1.6" href="#section-8.1.6">8.1.6</a>.  Keepalive</span>

   It is anticipated that media streams in SIPREC may exist in an
   inactive state for extended periods of time for any of a number of
   valid reasons.  In order for the bindings and any pinholes in
   NATs/firewalls to remain active during such intervals, it is
   RECOMMENDED that the SRC, SRS, and recording-aware UAs follow the
   keepalive procedure recommended in "Application Mechanism for Keeping
   Alive the NAT Mappings Associated with RTP / RTP Control Protocol
   (RTCP) Flows" [<a href="./rfc6263" title="&quot;Application Mechanism for Keeping Alive the NAT Mappings Associated with RTP / RTP Control Protocol (RTCP) Flows&quot;">RFC6263</a>] for all RTP media streams.



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<span class="h4"><a class="selflink" id="section-8.1.7" href="#section-8.1.7">8.1.7</a>.  RTCP Feedback Messages</span>

   "Codec Control Messages in the RTP Audio-Visual Profile with Feedback
   (AVPF)" [<a href="./rfc5104" title="&quot;Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF)&quot;">RFC5104</a>] specifies extensions to the messages defined in
   AVPF [<a href="./rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>].  Support for and proper usage of these messages are
   important to SRC, SRS, and recording-aware UA implementations.  Note
   that these messages are applicable only when using the AVPF or SAVPF
   RTP profiles.

<span class="h5"><a class="selflink" id="section-8.1.7.1" href="#section-8.1.7.1">8.1.7.1</a>.  Full Intra Request</span>

   A Full Intra Request (FIR) command, when received by the designated
   media sender, requires that the media sender send a decoder refresh
   point at the earliest opportunity.  Using a decoder refresh point
   implies refraining from using any picture sent prior to that point as
   a reference for the encoding process of any subsequent picture sent
   in the stream.

   Decoder refresh points, especially Intra or Instantaneous Decoding
   Refresh (IDR) pictures for H.264 video codecs, are in general several
   times larger in size than predicted pictures.  Thus, in scenarios in
   which the available bit rate is small, the use of a decoder refresh
   point implies a delay that is significantly longer than the typical
   picture duration.

<span class="h6"><a class="selflink" id="section-8.1.7.1.1" href="#section-8.1.7.1.1">8.1.7.1.1</a>.  Deprecated Usage of SIP INFO Instead of FIR</span>

   "XML Schema for Media Control" [<a href="./rfc5168" title="&quot;XML Schema for Media Control&quot;">RFC5168</a>] defines an Extensible Markup
   Language (XML) Schema for video fast update.  Implementations are
   discouraged from using the method described in [<a href="./rfc5168" title="&quot;XML Schema for Media Control&quot;">RFC5168</a>], except for
   purposes of backward compatibility.  Implementations SHOULD use FIR
   messages instead.

   To make sure that a common mechanism exists between the SRC and SRS,
   the SRS MUST support both mechanisms (FIR and SIP INFO), using FIR
   messages when negotiated successfully with the SRC and using SIP INFO
   otherwise.

<span class="h5"><a class="selflink" id="section-8.1.7.2" href="#section-8.1.7.2">8.1.7.2</a>.  Picture Loss Indication</span>

   Picture Loss Indication (PLI), as defined in [<a href="./rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>], informs the
   encoder of the loss of an undefined amount of coded video data
   belonging to one or more pictures.  [<a href="./rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>] recommends using PLI
   instead of FIR messages to recover from errors.  FIR is appropriate
   only in situations where not sending a decoder refresh point would
   render the video unusable for the users.  Examples where sending FIR
   messages is appropriate include a multipoint conference when a new




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   user joins the conference and no regular decoder refresh point
   interval is established, and a video-switching Multipoint Control
   Unit (MCU) that changes streams.

   Appropriate use of PLI and FIR is important to ensure, with minimum
   overhead, that the recorded video is usable (e.g., the necessary
   reference frames exist for a player to render the recorded video).

<span class="h5"><a class="selflink" id="section-8.1.7.3" href="#section-8.1.7.3">8.1.7.3</a>.  Temporary Maximum Media Stream Bit Rate Request</span>

   A receiver, translator, or mixer uses the Temporary Maximum Media
   Stream Bit Rate Request (TMMBR) [<a href="./rfc5104" title="&quot;Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF)&quot;">RFC5104</a>] to request a sender to
   limit the maximum bit rate for a media stream to the provided value.
   Appropriate use of TMMBR facilitates rapid adaptation to changes in
   available bandwidth.

<span class="h6"><a class="selflink" id="section-8.1.7.3.1" href="#section-8.1.7.3.1">8.1.7.3.1</a>.  Renegotiation of SDP Bandwidth Attribute</span>

   If it is likely that the new value indicated by TMMBR will be valid
   for the remainder of the session, the TMMBR sender is expected to
   perform a renegotiation of the session upper limit using the session
   signaling protocol.  Therefore, for SIPREC, implementations are
   RECOMMENDED to use TMMBR for temporary changes and renegotiation of
   bandwidth via SDP offer/answer for more permanent changes.

<span class="h4"><a class="selflink" id="section-8.1.8" href="#section-8.1.8">8.1.8</a>.  Symmetric RTP/RTCP for Sending and Receiving</span>

   Within an SDP offer/answer exchange, RTP entities choose the RTP and
   RTCP transport addresses (i.e., IP addresses and port numbers) on
   which to receive packets.  When sending packets, the RTP entities may
   use the same source port or a different source port than those
   signaled for receiving packets.  When the transport address used to
   send and receive RTP is the same, it is termed "symmetric RTP"
   [<a href="./rfc4961" title="&quot;Symmetric RTP / RTP Control Protocol (RTCP)&quot;">RFC4961</a>].  Likewise, when the transport address used to send and
   receive RTCP is the same, it is termed "symmetric RTCP" [<a href="./rfc4961" title="&quot;Symmetric RTP / RTP Control Protocol (RTCP)&quot;">RFC4961</a>].

   When sending RTP, the use of symmetric RTP is REQUIRED.  When sending
   RTCP, the use of symmetric RTCP is REQUIRED.  Although an SRS will
   not normally send RTP, it will send RTCP as well as receive RTP and
   RTCP.  Likewise, although an SRC will not normally receive RTP from
   the SRS, it will receive RTCP as well as send RTP and RTCP.

      Note: Symmetric RTP and symmetric RTCP are different from RTP/RTCP
      multiplexing [<a href="./rfc5761" title="&quot;Multiplexing RTP Data and Control Packets on a Single Port&quot;">RFC5761</a>].







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<span class="h3"><a class="selflink" id="section-8.2" href="#section-8.2">8.2</a>.  Roles</span>

   An SRC has the task of gathering media from the various UAs in one or
   more CSs and forwarding the information to the SRS within the context
   of a corresponding RS.  There are numerous ways in which an SRC may
   do this, including, but not limited to, appearing as a UA within a
   CS, or as a B2BUA between UAs within a CS.

                    (Recording Session)   +---------+
                  +------------SIP-------&gt;|         |
                  |  +------RTP/RTCP-----&gt;|   SRS   |
                  |  |    +-- Metadata --&gt;|         |
                  |  |    |               +---------+
                  v  v    |
                 +---------+
                 |   SRC   |
                 |---------| (Communication Session) +---------+
                 |         |&lt;----------SIP----------&gt;|         |
                 |  UA-A   |                         |  UA-B   |
                 |         |&lt;-------RTP/RTCP--------&gt;|         |
                 +---------+                         +---------+

                            Figure 8: UA as SRC


                                   (Recording Session)   +---------+
                                 +------------SIP-------&gt;|         |
                                 |  +------RTP/RTCP-----&gt;|   SRS   |
                                 |  |    +-- Metadata --&gt;|         |
                                 |  |    |               +---------+
                                 v  v    |
                                +---------+
                                |   SRC   |
       +---------+              |---------|              +---------+
       |         |&lt;----SIP-----&gt;|         |&lt;----SIP-----&gt;|         |
       |  UA-A   |              |  B2BUA  |              |  UA-B   |
       |         |&lt;--RTP/RTCP--&gt;|         |&lt;--RTP/RTCP--&gt;|         |
       +---------+              +---------+              +---------+
             |_______________________________________________|
                          (Communication Session)

                          Figure 9: B2BUA as SRC

   The following subsections define a set of roles an SRC may choose to
   play, based on its position with respect to a UA within a CS, and an
   SRS within an RS.  A CS and a corresponding RS are independent
   sessions; therefore, an SRC may play a different role within a CS
   than it does within the corresponding RS.



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<span class="h4"><a class="selflink" id="section-8.2.1" href="#section-8.2.1">8.2.1</a>.  SRC Acting as an RTP Translator</span>

   The SRC may act as a translator, as defined in [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>].  A defining
   characteristic of a translator is that it forwards RTP packets with
   their SSRC identifier intact.  There are two types of translators:
   one that simply forwards, and another that performs transcoding
   (e.g., from one codec to another) in addition to forwarding.

<span class="h5"><a class="selflink" id="section-8.2.1.1" href="#section-8.2.1.1">8.2.1.1</a>.  Forwarding Translator</span>

   When acting as a forwarding translator, RTP received as separate
   streams from different sources (e.g., from different UAs with
   different SSRCs) cannot be mixed by the SRC and MUST be sent
   separately to the SRS.  All RTCP reports MUST be passed by the SRC
   between the UAs and the SRS, such that the UAs and SRS are able to
   detect any SSRC collisions.

   RTCP Sender Reports generated by a UA sending a stream MUST be
   forwarded to the SRS.  RTCP Receiver Reports generated by the SRS
   MUST be forwarded to the relevant UA.

   UAs may receive multiple sets of RTCP Receiver Reports -- one or more
   from other UAs participating in the CS, and one from the SRS
   participating in the RS.  A UA SHOULD process the RTCP Receiver
   Reports from the SRS if it is recording aware.

   If SRTP is used on both the CS and the RS, decryption and/or
   re-encryption may occur.  For example, if different keys are used, it
   will occur.  If the same keys are used, it need not occur.
   <a href="#section-12">Section 12</a> provides additional information on SRTP and keying
   mechanisms.

   If packet loss occurs, either from the UA to the SRC or from the SRC
   to the SRS, the SRS SHOULD detect and attempt to recover from the
   loss.  The SRC does not play a role in this, other than forwarding
   the associated RTP and RTCP packets.

<span class="h5"><a class="selflink" id="section-8.2.1.2" href="#section-8.2.1.2">8.2.1.2</a>.  Transcoding Translator</span>

   When acting as a transcoding translator, an SRC MAY perform
   transcoding (e.g., from one codec to another), and this may result in
   a different rate of packets between what the SRC receives on the CS
   and what the SRC sends on the RS.  As when acting as a forwarding
   translator, RTP received as separate streams from different sources
   (e.g., from different UAs with different SSRCs) cannot be mixed by
   the SRC and MUST be sent separately to the SRS.  All RTCP reports
   MUST be passed by the SRC between the UAs and the SRS, such that the
   UAs and SRS are able to detect any SSRC collisions.



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   RTCP Sender Reports generated by a UA sending a stream MUST be
   forwarded to the SRS.  RTCP Receiver Reports generated by the SRS
   MUST be forwarded to the relevant UA.  The SRC may need to manipulate
   the RTCP Receiver Reports to take into account any transcoding that
   has taken place.

   UAs may receive multiple sets of RTCP Receiver Reports -- one or more
   from other UAs participating in the CS, and one from the SRS
   participating in the RS.  A recording-aware UA SHOULD be prepared to
   process the RTCP Receiver Reports from the SRS, whereas a recording-
   unaware UA may discard such RTCP packets as irrelevant.

   If SRTP is used on both the CS and the RS, decryption and/or
   re-encryption may occur.  For example, if different keys are used, it
   will occur.  If the same keys are used, it need not occur.
   <a href="#section-12">Section 12</a> provides additional information on SRTP and keying
   mechanisms.

   If packet loss occurs, either from the UA to the SRC or from the SRC
   to the SRS, the SRS SHOULD detect and attempt to recover from the
   loss.  The SRC does not play a role in this, other than forwarding
   the associated RTP and RTCP packets.

<span class="h4"><a class="selflink" id="section-8.2.2" href="#section-8.2.2">8.2.2</a>.  SRC Acting as an RTP Mixer</span>

   In the case of the SRC acting as an RTP mixer, as defined in
   [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>], the SRC combines RTP streams from different UAs and sends
   them towards the SRS using its own SSRC.  The SSRCs from the
   contributing UA SHOULD be conveyed as CSRC identifiers within this
   stream.  The SRC may make timing adjustments among the received
   streams and generate its own timing on the stream sent to the SRS.
   Optionally, an SRC acting as a mixer can perform transcoding and can
   even cope with different codings received from different UAs.  RTCP
   Sender Reports and Receiver Reports are not forwarded by an SRC
   acting as a mixer, but there are requirements for forwarding RTCP
   Source Description (SDES) packets.  The SRC generates its own RTCP
   Sender Reports and Receiver Reports toward the associated UAs
   and SRS.

   The use of SRTP between the SRC and the SRS for the RS is independent
   of the use of SRTP between the UAs and the SRC for the CS.
   <a href="#section-12">Section 12</a> provides additional information on SRTP and keying
   mechanisms.

   If packet loss occurs from the UA to the SRC, the SRC SHOULD detect
   and attempt to recover from the loss.  If packet loss occurs from
   the SRC to the SRS, the SRS SHOULD detect and attempt to recover from
   the loss.



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<span class="h4"><a class="selflink" id="section-8.2.3" href="#section-8.2.3">8.2.3</a>.  SRC Acting as an RTP Endpoint</span>

   The case of the SRC acting as an RTP endpoint, as defined in
   [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>], is similar to the mixer case, except that the RTP session
   between the SRC and the SRS is considered completely independent from
   the RTP session that is part of the CS.  The SRC can, but need not,
   mix RTP streams from different participants prior to sending to the
   SRS.  RTCP between the SRC and the SRS is completely independent of
   RTCP on the CS.

   The use of SRTP between the SRC and the SRS for the RS is independent
   of the use of SRTP between the UAs and SRC for the CS.  <a href="#section-12">Section 12</a>
   provides additional information on SRTP and keying mechanisms.

   If packet loss occurs from the UA to the SRC, the SRC SHOULD detect
   and attempt to recover from the loss.  If packet loss occurs from
   the SRC to the SRS, the SRS SHOULD detect and attempt to recover from
   the loss.

<span class="h3"><a class="selflink" id="section-8.3" href="#section-8.3">8.3</a>.  RTP Session Usage by SRC</span>

   There are multiple ways that an SRC may choose to deliver recorded
   media to an SRS.  In some cases, it may use a single RTP session for
   all media within the RS, whereas in others it may use multiple RTP
   sessions.  The following subsections provide examples of basic RTP
   session usage by the SRC, including a discussion of how the RTP
   constructs and mechanisms covered previously are used.  An SRC may
   choose to use one or more of the RTP session usages within a single
   RS.  For the purpose of base interoperability between SRC and SRS, an
   SRC MUST support separate m-lines in SDP, one per CS media direction.
   The set of RTP session usages described is not meant to be
   exhaustive.

<span class="h4"><a class="selflink" id="section-8.3.1" href="#section-8.3.1">8.3.1</a>.  SRC Using Multiple m-lines</span>

   When using multiple m-lines, an SRC includes each m-line in an SDP
   offer to the SRS.  The SDP answer from the SRS MUST include all
   m-lines, with any rejected m-lines indicated with a zero port, per
   [<a href="./rfc3264" title="&quot;An Offer/Answer Model with Session Description Protocol (SDP)&quot;">RFC3264</a>].  Having received the answer, the SRC starts sending media
   to the SRS as indicated in the answer.  Alternatively, if the SRC
   deems the level of support indicated in the answer to be
   unacceptable, it may initiate another SDP offer/answer exchange in
   which an alternative RTP session usage is negotiated.








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   In order to preserve the mapping of media to participant within the
   CSs in the RS, the SRC SHOULD map each unique CNAME within the CSs to
   a unique CNAME within the RS.  Additionally, the SRC SHOULD map each
   unique combination of CNAME/SSRC within the CSs to a unique
   CNAME/SSRC within the RS.  In doing so, the SRC may act as an
   RTP translator or as an RTP endpoint.

   Figure 10 illustrates a case in which each UA represents a
   participant contributing two RTP sessions (e.g., one for audio and
   one for video), each with a single SSRC.  The SRC acts as an RTP
   translator and delivers the media to the SRS using four RTP sessions,
   each with a single SSRC.  The CNAME and SSRC values used by the UAs
   within their media streams are preserved in the media streams from
   the SRC to the SRS.

                                                        +---------+
                                +------------SSRC Aa---&gt;|         |
                                |  + --------SSRC Av---&gt;|         |
                                |  |  +------SSRC Ba---&gt;|   SRS   |
                                |  |  |  +---SSRC Bv---&gt;|         |
                                |  |  |  |              +---------+
                                |  |  |  |
                                |  |  |  |
       +---------+             +----------+             +---------+
       |         |---SSRC Aa--&gt;|   SRC    |&lt;--SSRC Ba---|         |
       |  UA-A   |             |(CNAME-A, |             |  UA-B   |
       |(CNAME-A)|---SSRC Av--&gt;| CNAME-B) |&lt;--SSRC Bv---|(CNAME-B)|
       +---------+             +----------+             +---------+

                   Figure 10: SRC Using Multiple m-lines

<span class="h4"><a class="selflink" id="section-8.3.2" href="#section-8.3.2">8.3.2</a>.  SRC Using Mixing</span>

   When using mixing, the SRC combines RTP streams from different
   participants and sends them towards the SRS using its own SSRC.  The
   SSRCs from the contributing participants SHOULD be conveyed as CSRC
   identifiers.  The SRC includes one m-line for each RTP session in an
   SDP offer to the SRS.  The SDP answer from the SRS MUST include all
   m-lines, with any rejected m-lines indicated with a zero port, per
   [<a href="./rfc3264" title="&quot;An Offer/Answer Model with Session Description Protocol (SDP)&quot;">RFC3264</a>].  Having received the answer, the SRC starts sending media
   to the SRS as indicated in the answer.

   In order to preserve the mapping of media to participant within the
   CSs in the RS, the SRC SHOULD map each unique CNAME within the CSs to
   a unique CNAME within the RS.  Additionally, the SRC SHOULD map each
   unique combination of CNAME/SSRC within the CSs to a unique





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   CNAME/SSRC within the RS.  The SRC MUST avoid SSRC collisions,
   rewriting SSRCs if necessary when used as CSRCs in the RS.  In
   doing so, the SRC acts as an RTP mixer.

   In the event that the SRS does not support this usage of CSRC values,
   it relies entirely on the SIPREC metadata to determine the
   participants included within each mixed stream.

   Figure 11 illustrates a case in which each UA represents a
   participant contributing two RTP sessions (e.g., one for audio and
   one for video), each with a single SSRC.  The SRC acts as an RTP
   mixer and delivers the media to the SRS using two RTP sessions,
   mixing media from each participant into a single RTP session
   containing a single SSRC and two CSRCs.

                                          SSRC Sa       +---------+
                                  +-------CSRC Aa,Ba---&gt;|         |
                                  |                     |         |
                                  |       SSRC Sv       |   SRS   |
                                  |   +---CSRC Av,Bv---&gt;|         |
                                  |   |                 +---------+
                                  |   |
                               +----------+
       +---------+             |   SRC    |             +---------+
       |         |---SSRC Aa--&gt;|(CNAME-S, |&lt;--SSRC Ba---|         |
       |  UA-A   |             | CNAME-A, |             |  UA-B   |
       |(CNAME-A)|---SSRC Av--&gt;| CNAME-B) |&lt;--SSRC Bv---|(CNAME-B)|
       +---------+             +----------+             +---------+

                        Figure 11: SRC Using Mixing

<span class="h3"><a class="selflink" id="section-8.4" href="#section-8.4">8.4</a>.  RTP Session Usage by SRS</span>

   An SRS that supports recording an audio CS MUST support SRC usage of
   separate audio m-lines in SDP, one per CS media direction.  An SRS
   that supports recording a video CS MUST support SRC usage of separate
   video m-lines in SDP, one per CS media direction.  Therefore, for an
   SRS supporting a typical audio call, the SRS has to support receiving
   at least two audio m-lines.  For an SRS supporting a typical audio
   and video call, the SRS has to support receiving at least four total
   m-lines in the SDP -- two audio m-lines and two video m-lines.

   These requirements allow an SRS to be implemented that supports video
   only, without requiring support for audio recording.  They also allow
   an SRS to be implemented that supports recording only one direction
   of one stream in a CS -- for example, an SRS designed to record
   security monitoring cameras that only send (not receive) video
   without any audio.  These requirements were not written to prevent



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   other modes from being implemented and used, such as using a single
   m-line and mixing the separate audio streams together.  Rather, the
   requirements were written to provide a common base mode to implement
   for the sake of interoperability.  It is important to note that an
   SRS implementation supporting the common base mode may not record all
   media streams in a CS if a participant supports more than one m-line
   in a video call, such as one for camera and one for presentation.
   SRS implementations may support other modes as well, but they have to
   at least support the modes discussed above, such that they
   interoperate in the common base mode for basic interoperability.

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

   Some metadata attributes are contained in SDP, and others are
   contained in a new content type called "application/rs-metadata".
   The format of the metadata is described as part of the mechanism in
   [<a href="./rfc7865" title="&quot;Session Initiation Protocol (SIP) Recording Metadata&quot;">RFC7865</a>].  A new "disposition-type" of Content-Disposition is
   defined for the purpose of carrying metadata.  The value is
   "recording-session", which indicates that the
   "application/rs-metadata" content contains metadata to be handled by
   the SRS.

<span class="h3"><a class="selflink" id="section-9.1" href="#section-9.1">9.1</a>.  Procedures at the SRC</span>

   The SRC MUST send metadata to the SRS in an RS.  The SRC SHOULD send
   metadata as soon as it becomes available and whenever it changes.
   Cases in which an SRC may be justified in waiting temporarily before
   sending metadata include:

   o  waiting for a previous metadata exchange to complete (i.e., the
      SRC cannot send another SDP offer until the previous offer/answer
      completes and may also prefer not to send an UPDATE during this
      time).

   o  constraining the signaling rate on the RS.

   o  sending metadata when key events occur, rather than for every
      event that has any impact on metadata.

   The SRC may also be configured to suppress certain metadata out of
   concern for privacy or perceived lack of need for it to be included
   in the recording.

   Metadata sent by the SRC is categorized as either a full metadata
   snapshot or a partial update.  A full metadata snapshot describes all
   metadata associated with the RS.  The SRC MAY send a full metadata
   snapshot at any time.  The SRC MAY send a partial update only if a
   full metadata snapshot has been sent previously.



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   The SRC MAY send metadata (either a full metadata snapshot or a
   partial update) in an INVITE request, an UPDATE request [<a href="./rfc3311" title="&quot;The Session Initiation Protocol (SIP) UPDATE Method&quot;">RFC3311</a>], or
   a 200 response to an offerless INVITE from the SRS.  If the metadata
   contains a reference to any SDP labels, the request containing the
   metadata MUST also contain an SDP offer that defines those labels.

   When a SIP message contains both an SDP offer and metadata, the
   request body MUST have content type "multipart/mixed", with one
   subordinate body part containing the SDP offer and another containing
   the metadata.  When a SIP message contains only an SDP offer or
   metadata, the "multipart/mixed" container is optional.

   The SRC SHOULD include a full metadata snapshot in the initial INVITE
   request establishing the RS.  If metadata is not yet available (e.g.,
   an RS established in the absence of a CS), the SRC SHOULD send a full
   metadata snapshot as soon as metadata becomes available.

   If the SRC receives a snapshot request from the SRS, it MUST
   immediately send a full metadata snapshot.
































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   Figure 12 illustrates an example of a full metadata snapshot sent by
   the SRC in the initial INVITE request:

       INVITE sip:recorder@example.com SIP/2.0
       Via: SIP/2.0/TCP src.example.com;branch=z9hG4bKdf6b622b648d9
       From: &lt;sip:2000@example.com&gt;;tag=35e195d2-947d-4585-946f-09839247
       To: &lt;sip:recorder@example.com&gt;
       Call-ID: d253c800-b0d1ea39-4a7dd-3f0e20a
       CSeq: 101 INVITE
       Max-Forwards: 70
       Require: siprec
       Accept: application/sdp, application/rs-metadata
       Contact: &lt;sip:2000@src.example.com&gt;;+sip.src
       Content-Type: multipart/mixed;boundary=foobar
       Content-Length: [length]

       --foobar
       Content-Type: application/sdp

       v=0
       o=SRS 2890844526 2890844526 IN IP4 198.51.100.1
       s=-
       c=IN IP4 198.51.100.1
       t=0 0
       m=audio 12240 RTP/AVP 0 4 8
       a=sendonly
       a=label:1

       --foobar
       Content-Type: application/rs-metadata
       Content-Disposition: recording-session

       [metadata content]

        Figure 12: Sample INVITE Request for the Recording Session

<span class="h3"><a class="selflink" id="section-9.2" href="#section-9.2">9.2</a>.  Procedures at the SRS</span>

   The SRS receives metadata updates from the SRC in INVITE and UPDATE
   requests.  Since the SRC can send partial updates based on the
   previous update, the SRS needs to keep track of the sequence of
   updates from the SRC.

   In the case of an internal failure at the SRS, the SRS may fail to
   recognize a partial update from the SRC.  The SRS may be able to
   recover from the internal failure by requesting a full metadata
   snapshot from the SRC.  Certain errors, such as syntax errors or
   semantic errors in the metadata information, are likely caused by an



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   error on the SRC side, and it is likely that the same error will
   occur again even when a full metadata snapshot is requested.  In
   order to avoid repeating the same error, the SRS can simply terminate
   the RS when a syntax error or semantic error is detected in the
   metadata.

   The SRS MAY explicitly request a full metadata snapshot by sending an
   UPDATE request.  This request MUST contain a body with
   Content-Disposition type "recording-session" and MUST NOT contain an
   SDP body.  The SRS MUST NOT request a full metadata snapshot in an
   UPDATE response or in any other SIP transaction.  The format of the
   content is "application/rs-metadata", and the body is an XML
   document, the format of which is defined in [<a href="./rfc7865" title="&quot;Session Initiation Protocol (SIP) Recording Metadata&quot;">RFC7865</a>].  Figure 13
   shows an example:

     UPDATE sip:2000@src.example.com SIP/2.0
     Via: SIP/2.0/UDP srs.example.com;branch=z9hG4bKdf6b622b648d9
     To: &lt;sip:2000@example.com&gt;;tag=35e195d2-947d-4585-946f-098392474
     From: &lt;sip:recorder@example.com&gt;;tag=1234567890
     Call-ID: d253c800-b0d1ea39-4a7dd-3f0e20a
     CSeq: 1 UPDATE
     Max-Forwards: 70
     Require: siprec
     Contact: &lt;sip:recorder@srs.example.com&gt;;+sip.srs
     Accept: application/sdp, application/rs-metadata
     Content-Disposition: recording-session
     Content-Type: application/rs-metadata
     Content-Length: [length]

     &lt;?xml version="1.0" encoding="UTF-8"?&gt;
       &lt;requestsnapshot xmlns='urn:ietf:params:xml:ns:recording:1'&gt;
         &lt;requestreason xml:lang="it"&gt;SRS internal error&lt;/requestreason&gt;
       &lt;/requestsnapshot&gt;

                        Figure 13: Metadata Request

   Note that UPDATE was chosen for the SRS to request a metadata
   snapshot, because it can be sent regardless of the state of the
   dialog.  This was seen as better than requiring support for both
   UPDATE and re-INVITE messages for this operation.

   When the SRC receives a request for a metadata snapshot, it MUST
   immediately provide a full metadata snapshot in a separate INVITE or
   UPDATE transaction.  Any subsequent partial updates will not be
   dependent on any metadata sent prior to this full metadata snapshot.






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   The metadata received by the SRS can contain ID elements used to
   cross-reference one element to another.  An element containing the
   definition of an ID and an element containing a reference to that ID
   will often be received from the same SRC.  It is also valid for those
   elements to be received from different SRCs -- for example, when each
   endpoint in the same CS acts as an SRC to record the call and a
   common ID refers to the same CS.  The SRS MUST NOT consider this an
   error.

<span class="h2"><a class="selflink" id="section-10" href="#section-10">10</a>.  Persistent Recording</span>

   Persistent recording is a specific use case addressing REQ-005 in
   [<a href="./rfc6341" title="&quot;Use Cases and Requirements for SIP-Based Media Recording (SIPREC)&quot;">RFC6341</a>], where an RS can be established in the absence of a CS.
   The SRC continuously records media in an RS to the SRS even in the
   absence of a CS for all UAs that are part of persistent recording.
   By allocating recorded streams and continuously sending recorded
   media to the SRS, the SRC does not have to prepare new recorded
   streams with a new SDP offer when a new CS is created and also does
   not impact the timing of the CS.  The SRC only needs to update the
   metadata when new CSs are created.

   When there is no CS running on the devices with persistent recording,
   there is no recorded media to stream from the SRC to the SRS.  In
   certain environments where a Network Address Translator (NAT) is
   used, a minimum amount of flow activity is typically required to
   maintain the NAT binding for each port opened.  Agents that support
   Interactive Connectivity Establishment (ICE) solve this problem.  For
   non-ICE agents, in order not to lose the NAT bindings for the
   RTP/RTCP ports opened for the recorded streams, the SRC and SRS
   SHOULD follow the recommendations provided in [<a href="./rfc6263" title="&quot;Application Mechanism for Keeping Alive the NAT Mappings Associated with RTP / RTP Control Protocol (RTCP) Flows&quot;">RFC6263</a>] to maintain
   the NAT bindings.




















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<span class="h2"><a class="selflink" id="section-11" href="#section-11">11</a>.  IANA Considerations</span>

<span class="h3"><a class="selflink" id="section-11.1" href="#section-11.1">11.1</a>.  Registration of Option Tags</span>

   This specification registers two option tags.  The required
   information for this registration, as specified in [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>], is as
   follows.

<span class="h4"><a class="selflink" id="section-11.1.1" href="#section-11.1.1">11.1.1</a>.  "siprec" Option Tag</span>

   Name:  siprec

   Description:  This option tag is for identifying that the SIP session
      is for the purpose of an RS.  This is typically not used in a
      Supported header.  When present in a Require header in a request,
      it indicates that the UA is either an SRC or SRS capable of
      handling an RS.

<span class="h4"><a class="selflink" id="section-11.1.2" href="#section-11.1.2">11.1.2</a>.  "record-aware" Option Tag</span>

   Name:  record-aware

   Description:  This option tag is to indicate the ability of the UA to
      receive recording indicators in media-level or session-level SDP.
      When present in a Supported header, it indicates that the UA can
      receive recording indicators in media-level or session-level SDP.

<span class="h3"><a class="selflink" id="section-11.2" href="#section-11.2">11.2</a>.  Registration of Media Feature Tags</span>

   This document registers two new media feature tags in the SIP tree
   per the process defined in [<a href="./rfc2506" title="&quot;Media Feature Tag Registration Procedure&quot;">RFC2506</a>] and [<a href="./rfc3840" title="&quot;Indicating User Agent Capabilities in the Session Initiation Protocol (SIP)&quot;">RFC3840</a>].

<span class="h4"><a class="selflink" id="section-11.2.1" href="#section-11.2.1">11.2.1</a>.  Feature Tag for the SRC</span>

   Media feature tag name:  sip.src

   ASN.1 Identifier:  1.3.6.1.8.4.27

   Summary of the media feature indicated by this tag:  This feature tag
      indicates that the UA is a Session Recording Client for the
      purpose of an RS.

   Values appropriate for use with this feature tag:  boolean

   The feature tag is intended primarily for use in the following
      applications, protocols, services, or negotiation mechanisms:
      This feature tag is only useful for an RS.




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   Examples of typical use:  Routing the request to a Session Recording
      Server.

   Security Considerations:  Security considerations for this media
      feature tag are discussed in <a href="./rfc3840#section-11.1">Section&nbsp;11.1 of RFC 3840</a>.

<span class="h4"><a class="selflink" id="section-11.2.2" href="#section-11.2.2">11.2.2</a>.  Feature Tag for the SRS</span>

   Media feature tag name:  sip.srs

   ASN.1 Identifier:  1.3.6.1.8.4.28

   Summary of the media feature indicated by this tag:  This feature tag
      indicates that the UA is a Session Recording Server for the
      purpose of an RS.

   Values appropriate for use with this feature tag:  boolean

   The feature tag is intended primarily for use in the following
      applications, protocols, services, or negotiation mechanisms:
      This feature tag is only useful for an RS.

   Examples of typical use:  Routing the request to a Session Recording
      Client.

   Security Considerations:  Security considerations for this media
      feature tag are discussed in <a href="./rfc3840#section-11.1">Section&nbsp;11.1 of RFC 3840</a>.

<span class="h3"><a class="selflink" id="section-11.3" href="#section-11.3">11.3</a>.  New Content-Disposition Parameter Registrations</span>

   This document registers a new "disposition-type" value in the
   Content-Disposition header: recording-session.

   recording-session:  The body describes either

      *  metadata about the RS

         or

      *  the reason for the metadata snapshot request

      as determined by the MIME value indicated in the Content-Type.









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<span class="h3"><a class="selflink" id="section-11.4" href="#section-11.4">11.4</a>.  SDP Attributes</span>

   This document registers the following new SDP attributes.

<span class="h4"><a class="selflink" id="section-11.4.1" href="#section-11.4.1">11.4.1</a>.  "record" SDP Attribute</span>

   Contact names:
      Leon Portman, leon.portman@nice.com;
      Henry Lum, henry.lum@genesyslab.com

   Attribute name: record

   Long-form attribute name: Recording Indication

   Type of attribute: session level or media level

   Subject to charset: no

   This attribute provides the recording indication for the session or
   media stream.

   Allowed attribute values: on, off, paused

<span class="h4"><a class="selflink" id="section-11.4.2" href="#section-11.4.2">11.4.2</a>.  "recordpref" SDP Attribute</span>

   Contact names:
      Leon Portman, leon.portman@nice.com;
      Henry Lum, henry.lum@genesyslab.com

   Attribute name: recordpref

   Long-form attribute name: Recording Preference

   Type of attribute: session level or media level

   Subject to charset: no

   This attribute provides the recording preference for the session or
   media stream.

   Allowed attribute values: on, off, pause, nopreference










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<span class="h2"><a class="selflink" id="section-12" href="#section-12">12</a>.  Security Considerations</span>

   The RS is fundamentally a standard SIP dialog [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>]; therefore,
   the RS can reuse any of the existing SIP security mechanisms
   available for securing the session signaling, the recorded media, and
   the metadata.  The use cases and requirements document [<a href="./rfc6341" title="&quot;Use Cases and Requirements for SIP-Based Media Recording (SIPREC)&quot;">RFC6341</a>]
   outlines the general security considerations, and this document
   describes specific security recommendations.

   The SRC and SRS MUST support SIP with Transport Layer Security (TLS)
   version 1.2, SHOULD follow the best practices when using TLS as per
   [<a href="./rfc7525" title="&quot;Recommendations for Secure Use of Transport Layer Security (TLS) and Datagram Transport Layer Security (DTLS)&quot;">RFC7525</a>], and MAY use Session Initiation Protocol Secure (SIPS) with
   TLS as per [<a href="./rfc5630" title="&quot;The Use of the SIPS URI Scheme in the Session Initiation Protocol (SIP)&quot;">RFC5630</a>].  The RS MUST be at least as secure as the CS;
   this means using at least the same strength of cipher suite as the CS
   if the CS is secured.  For example, if the CS uses SIPS for signaling
   and RTP/SAVP for media, then the RS may not use SIP or plain RTP
   unless other equivalent security measures are in effect, since doing
   so would mean an effective security downgrade.  Examples of other
   potentially equivalent security mechanisms include mutually
   authenticated TLS for the RS signaling channel or an appropriately
   protected network path for the RS media component.

<span class="h3"><a class="selflink" id="section-12.1" href="#section-12.1">12.1</a>.  Authentication and Authorization</span>

   At the transport level, the RS uses TLS authentication to validate
   the authenticity of the SRC and SRS.  The SRC and SRS MUST implement
   TLS mutual authentication for establishing the RS.  Whether the
   SRC/SRS chooses to use TLS mutual authentication is a deployment
   decision.  In deployments where a UA acts as its own SRC, this
   requires that the UA have its own certificate as needed for TLS
   mutual authentication.  In deployments where the SRC and the SRS are
   in the same administrative domain and have some other means of
   assuring authenticity, the SRC and SRS may choose not to authenticate
   each other or to have the SRC authenticate the SRS only.  In
   deployments where the SRS can be hosted on a different administrative
   domain, it is important to perform mutual authentication to ensure
   the authenticity of both the SRC and the SRS before transmitting any
   recorded media.  The risk of not authenticating the SRS is that the
   recording may be sent to an entity other than the intended SRS,
   allowing a sensitive call recording to be received by an attacker.
   On the other hand, the risk of not authenticating the SRC is that an
   SRS will accept calls from an unknown SRC and allow potential forgery
   of call recordings.

   There may be scenarios in which the signaling between the SRC and SRS
   is not direct, e.g., a SIP proxy exists between the SRC and the SRS.
   In such scenarios, each hop is subject to the TLS mutual
   authentication constraint, and transitive trust at each hop is



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   utilized.  Additionally, an SRC or SRS may use other existing SIP
   mechanisms available, including, but not limited to, Digest
   authentication [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>], asserted identity [<a href="./rfc3325" title="&quot;Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks&quot;">RFC3325</a>], and connected
   identity [<a href="./rfc4916" title="&quot;Connected Identity in the Session Initiation Protocol (SIP)&quot;">RFC4916</a>].

   The SRS may have its own set of recording policies to authorize
   recording requests from the SRC.  The use of recording policies is
   outside the scope of the Session Recording Protocol.

<span class="h3"><a class="selflink" id="section-12.2" href="#section-12.2">12.2</a>.  RTP Handling</span>

   In many scenarios, it will be critical for the media transported
   between the SRC and the SRS to be protected.  Media encryption is an
   important element in the overall SIPREC solution; therefore, the SRC
   and the SRS MUST support RTP/SAVP [<a href="./rfc3711" title="&quot;The Secure Real-time Transport Protocol (SRTP)&quot;">RFC3711</a>] and RTP/SAVPF [<a href="./rfc5124" title="&quot;Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)&quot;">RFC5124</a>].
   RTP/SAVP and RTP/SAVPF provide media encryption, integrity
   protection, replay protection, and a limited form of source
   authentication.  They do not contain or require a specific keying
   mechanism.  At a minimum, the SRC and SRS MUST support the SDP
   security descriptions key negotiation mechanism [<a href="./rfc4568" title="&quot;Session Description Protocol (SDP) Security Descriptions for Media Streams&quot;">RFC4568</a>].  For cases
   in which Datagram Transport Layer Security for Secure RTP (DTLS-SRTP)
   is used to encrypt a CS media stream, an SRC may use SRTP Encrypted
   Key Transport (EKT) [<a href="#ref-EKT-SRTP" title="&quot;Encrypted Key Transport for Secure RTP&quot;">EKT-SRTP</a>] in order to use SRTP-SDES in the RS
   without needing to re-encrypt the media.

      Note: When using EKT in this manner, it is possible for
      participants in the CS to send traffic that appears to be from
      other participants and have this forwarded by the SRC to the SRS
      within the RS.  If this is a concern (e.g., the RS is intended for
      audit or compliance purposes), EKT is not an appropriate choice.

   When RTP/SAVP or RTP/SAVPF is used, an SRC can choose to use the same
   keys or different keys in the RS than those used in the CS.  Some
   SRCs are designed to simply replicate RTP packets from a CS media
   stream to the SRS, in which case the SRC will use the same key in the
   RS as the key used in the CS.  In this case, the SRC MUST secure the
   SDP containing the keying material in the RS with at least the same
   level of security as in the CS.  The risk of lowering the level of
   security in the RS is that it will effectively become a downgrade
   attack on the CS, since the same key is used for both the CS and
   the RS.

   SRCs that decrypt an encrypted CS media stream and re-encrypt it when
   sending it to the SRS MUST use a different key than what is used for
   the CS media stream, to ensure that it is not possible for someone
   who has the key for the CS media stream to access recorded data they





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   are not authorized to access.  In order to maintain a comparable
   level of security, the key used in the RS SHOULD be of equivalent
   strength to, or greater strength than, that used in the CS.

<span class="h3"><a class="selflink" id="section-12.3" href="#section-12.3">12.3</a>.  Metadata</span>

   Metadata contains sensitive information, such as the address of
   record of the participants and other extension data placed by the
   SRC.  It is essential to protect the content of the metadata in the
   RS.  Since metadata is a content type transmitted in SIP signaling,
   metadata SHOULD be protected at the transport level by SIPS/TLS.

<span class="h3"><a class="selflink" id="section-12.4" href="#section-12.4">12.4</a>.  Storage and Playback</span>

   While storage and playback of the call recording are beyond the scope
   of this document, it is worthwhile to mention here that it is also
   important for the recording storage and playback to provide a level
   of security that is comparable to the CS.  It would defeat the
   purpose of securing both the CS and the RS mentioned in the previous
   sections if the recording can be easily played back with a simple,
   unsecured HTTP interface without any form of authentication or
   authorization.

<span class="h2"><a class="selflink" id="section-13" href="#section-13">13</a>.  References</span>

<span class="h3"><a class="selflink" id="section-13.1" href="#section-13.1">13.1</a>.  Normative References</span>

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

   [<a id="ref-RFC2506">RFC2506</a>]  Holtman, K., Mutz, A., and T. Hardie, "Media Feature Tag
              Registration Procedure", <a href="https://www.rfc-editor.org/bcp/bcp31">BCP 31</a>, <a href="./rfc2506">RFC 2506</a>,
              DOI 10.17487/RFC2506, March 1999,
              &lt;<a href="http://www.rfc-editor.org/info/rfc2506">http://www.rfc-editor.org/info/rfc2506</a>&gt;.

   [<a id="ref-RFC3261">RFC3261</a>]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", <a href="./rfc3261">RFC 3261</a>,
              DOI 10.17487/RFC3261, June 2002,
              &lt;<a href="http://www.rfc-editor.org/info/rfc3261">http://www.rfc-editor.org/info/rfc3261</a>&gt;.

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




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

   [<a id="ref-RFC3840">RFC3840</a>]  Rosenberg, J., Schulzrinne, H., and P. Kyzivat,
              "Indicating User Agent Capabilities in the Session
              Initiation Protocol (SIP)", <a href="./rfc3840">RFC 3840</a>,
              DOI 10.17487/RFC3840, August 2004,
              &lt;<a href="http://www.rfc-editor.org/info/rfc3840">http://www.rfc-editor.org/info/rfc3840</a>&gt;.

   [<a id="ref-RFC4574">RFC4574</a>]  Levin, O. and G. Camarillo, "The Session Description
              Protocol (SDP) Label Attribute", <a href="./rfc4574">RFC 4574</a>,
              DOI 10.17487/RFC4574, August 2006,
              &lt;<a href="http://www.rfc-editor.org/info/rfc4574">http://www.rfc-editor.org/info/rfc4574</a>&gt;.

   [<a id="ref-RFC5234">RFC5234</a>]  Crocker, D., Ed., and P. Overell, "Augmented BNF for
              Syntax Specifications: ABNF", STD 68, <a href="./rfc5234">RFC 5234</a>,
              DOI 10.17487/RFC5234, January 2008,
              &lt;<a href="http://www.rfc-editor.org/info/rfc5234">http://www.rfc-editor.org/info/rfc5234</a>&gt;.

   [<a id="ref-RFC7245">RFC7245</a>]  Hutton, A., Ed., Portman, L., Ed., Jain, R., and K. Rehor,
              "An Architecture for Media Recording Using the Session
              Initiation Protocol", <a href="./rfc7245">RFC 7245</a>, DOI 10.17487/RFC7245,
              May 2014, &lt;<a href="http://www.rfc-editor.org/info/rfc7245">http://www.rfc-editor.org/info/rfc7245</a>&gt;.

   [<a id="ref-RFC7865">RFC7865</a>]  Ravindranath, R., Ravindran, P., and P. Kyzivat, "Session
              Initiation Protocol (SIP) Recording Metadata", <a href="./rfc7865">RFC 7865</a>,
              DOI 10.17487/RFC7865, May 2016,
              &lt;<a href="http://www.rfc-editor.org/info/rfc7865">http://www.rfc-editor.org/info/rfc7865</a>&gt;.

<span class="h3"><a class="selflink" id="section-13.2" href="#section-13.2">13.2</a>.  Informative References</span>

   [<a id="ref-EKT-SRTP">EKT-SRTP</a>] Mattsson, J., Ed., McGrew, D., Wing, D., and F. Andreasen,
              "Encrypted Key Transport for Secure RTP", Work in
              Progress, <a href="./draft-ietf-avtcore-srtp-ekt-03">draft-ietf-avtcore-srtp-ekt-03</a>, October 2014.

   [<a id="ref-RFC2804">RFC2804</a>]  IAB and IESG, "IETF Policy on Wiretapping", <a href="./rfc2804">RFC 2804</a>,
              DOI 10.17487/RFC2804, May 2000,
              &lt;<a href="http://www.rfc-editor.org/info/rfc2804">http://www.rfc-editor.org/info/rfc2804</a>&gt;.

   [<a id="ref-RFC3311">RFC3311</a>]  Rosenberg, J., "The Session Initiation Protocol (SIP)
              UPDATE Method", <a href="./rfc3311">RFC 3311</a>, DOI 10.17487/RFC3311,
              October 2002, &lt;<a href="http://www.rfc-editor.org/info/rfc3311">http://www.rfc-editor.org/info/rfc3311</a>&gt;.







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   [<a id="ref-RFC3325">RFC3325</a>]  Jennings, C., Peterson, J., and M. Watson, "Private
              Extensions to the Session Initiation Protocol (SIP) for
              Asserted Identity within Trusted Networks", <a href="./rfc3325">RFC 3325</a>,
              DOI 10.17487/RFC3325, November 2002,
              &lt;<a href="http://www.rfc-editor.org/info/rfc3325">http://www.rfc-editor.org/info/rfc3325</a>&gt;.

   [<a id="ref-RFC3551">RFC3551</a>]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
              Video Conferences with Minimal Control", STD 65, <a href="./rfc3551">RFC 3551</a>,
              DOI 10.17487/RFC3551, July 2003,
              &lt;<a href="http://www.rfc-editor.org/info/rfc3551">http://www.rfc-editor.org/info/rfc3551</a>&gt;.

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

   [<a id="ref-RFC4568">RFC4568</a>]  Andreasen, F., Baugher, M., and D. Wing, "Session
              Description Protocol (SDP) Security Descriptions for Media
              Streams", <a href="./rfc4568">RFC 4568</a>, DOI 10.17487/RFC4568, July 2006,
              &lt;<a href="http://www.rfc-editor.org/info/rfc4568">http://www.rfc-editor.org/info/rfc4568</a>&gt;.

   [<a id="ref-RFC4585">RFC4585</a>]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
              "Extended RTP Profile for Real-time Transport Control
              Protocol (RTCP)-Based Feedback (RTP/AVPF)", <a href="./rfc4585">RFC 4585</a>,
              DOI 10.17487/RFC4585, July 2006,
              &lt;<a href="http://www.rfc-editor.org/info/rfc4585">http://www.rfc-editor.org/info/rfc4585</a>&gt;.

   [<a id="ref-RFC4916">RFC4916</a>]  Elwell, J., "Connected Identity in the Session Initiation
              Protocol (SIP)", <a href="./rfc4916">RFC 4916</a>, DOI 10.17487/RFC4916,
              June 2007, &lt;<a href="http://www.rfc-editor.org/info/rfc4916">http://www.rfc-editor.org/info/rfc4916</a>&gt;.

   [<a id="ref-RFC4961">RFC4961</a>]  Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
              <a href="https://www.rfc-editor.org/bcp/bcp131">BCP 131</a>, <a href="./rfc4961">RFC 4961</a>, DOI 10.17487/RFC4961, July 2007,
              &lt;<a href="http://www.rfc-editor.org/info/rfc4961">http://www.rfc-editor.org/info/rfc4961</a>&gt;.

   [<a id="ref-RFC5104">RFC5104</a>]  Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
              "Codec Control Messages in the RTP Audio-Visual Profile
              with Feedback (AVPF)", <a href="./rfc5104">RFC 5104</a>, DOI 10.17487/RFC5104,
              February 2008, &lt;<a href="http://www.rfc-editor.org/info/rfc5104">http://www.rfc-editor.org/info/rfc5104</a>&gt;.

   [<a id="ref-RFC5124">RFC5124</a>]  Ott, J. and E. Carrara, "Extended Secure RTP Profile for
              Real-time Transport Control Protocol (RTCP)-Based Feedback
              (RTP/SAVPF)", <a href="./rfc5124">RFC 5124</a>, DOI 10.17487/RFC5124,
              February 2008, &lt;<a href="http://www.rfc-editor.org/info/rfc5124">http://www.rfc-editor.org/info/rfc5124</a>&gt;.

   [<a id="ref-RFC5168">RFC5168</a>]  Levin, O., Even, R., and P. Hagendorf, "XML Schema for
              Media Control", <a href="./rfc5168">RFC 5168</a>, DOI 10.17487/RFC5168,
              March 2008, &lt;<a href="http://www.rfc-editor.org/info/rfc5168">http://www.rfc-editor.org/info/rfc5168</a>&gt;.



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   [<a id="ref-RFC5630">RFC5630</a>]  Audet, F., "The Use of the SIPS URI Scheme in the Session
              Initiation Protocol (SIP)", <a href="./rfc5630">RFC 5630</a>,
              DOI 10.17487/RFC5630, October 2009,
              &lt;<a href="http://www.rfc-editor.org/info/rfc5630">http://www.rfc-editor.org/info/rfc5630</a>&gt;.

   [<a id="ref-RFC5761">RFC5761</a>]  Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
              Control Packets on a Single Port", <a href="./rfc5761">RFC 5761</a>,
              DOI 10.17487/RFC5761, April 2010,
              &lt;<a href="http://www.rfc-editor.org/info/rfc5761">http://www.rfc-editor.org/info/rfc5761</a>&gt;.

   [<a id="ref-RFC6263">RFC6263</a>]  Marjou, X. and A. Sollaud, "Application Mechanism for
              Keeping Alive the NAT Mappings Associated with RTP / RTP
              Control Protocol (RTCP) Flows", <a href="./rfc6263">RFC 6263</a>,
              DOI 10.17487/RFC6263, June 2011,
              &lt;<a href="http://www.rfc-editor.org/info/rfc6263">http://www.rfc-editor.org/info/rfc6263</a>&gt;.

   [<a id="ref-RFC6341">RFC6341</a>]  Rehor, K., Ed., Portman, L., Ed., Hutton, A., and R. Jain,
              "Use Cases and Requirements for SIP-Based Media Recording
              (SIPREC)", <a href="./rfc6341">RFC 6341</a>, DOI 10.17487/RFC6341, August 2011,
              &lt;<a href="http://www.rfc-editor.org/info/rfc6341">http://www.rfc-editor.org/info/rfc6341</a>&gt;.

   [<a id="ref-RFC7022">RFC7022</a>]  Begen, A., Perkins, C., Wing, D., and E. Rescorla,
              "Guidelines for Choosing RTP Control Protocol (RTCP)
              Canonical Names (CNAMEs)", <a href="./rfc7022">RFC 7022</a>, DOI 10.17487/RFC7022,
              September 2013, &lt;<a href="http://www.rfc-editor.org/info/rfc7022">http://www.rfc-editor.org/info/rfc7022</a>&gt;.

   [<a id="ref-RFC7525">RFC7525</a>]  Sheffer, Y., Holz, R., and P. Saint-Andre,
              "Recommendations for Secure Use of Transport Layer
              Security (TLS) and Datagram Transport Layer Security
              (DTLS)", <a href="https://www.rfc-editor.org/bcp/bcp195">BCP 195</a>, <a href="./rfc7525">RFC 7525</a>, DOI 10.17487/RFC7525,
              May 2015, &lt;<a href="http://www.rfc-editor.org/info/rfc7525">http://www.rfc-editor.org/info/rfc7525</a>&gt;.

Acknowledgements

   We want to thank John Elwell, Paul Kyzivat, Partharsarathi R, Ram
   Mohan R, Hadriel Kaplan, Adam Roach, Miguel Garcia, Thomas Stach,
   Muthu Perumal, Dan Wing, and Magnus Westerlund for their valuable
   comments and inputs to this document.













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Authors' Addresses

   Leon Portman
   NICE Systems
   22 Zarhin Street
   P.O. Box 690
   Ra'anana  4310602
   Israel

   Email: leon.portman@gmail.com


   Henry Lum (editor)
   Genesys
   1380 Rodick Road, Suite 201
   Markham, Ontario  L3R4G5
   Canada

   Email: henry.lum@genesyslab.com


   Charles Eckel
   Cisco
   170 West Tasman Drive
   San Jose, CA  95134
   United States

   Email: eckelcu@cisco.com


   Alan Johnston
   Illinois Institute of Technology
   Bellevue, WA
   United States

   Email: alan.b.johnston@gmail.com


   Andrew Hutton
   Unify
   Brickhill Street
   Milton Keynes  MK15 0DJ
   United Kingdom

   Email: andrew.hutton@unify.com






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