Session Recording Protocol
draft-ietf-siprec-protocol-10

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SIPREC                                                      L.P. Portman
Internet-Draft                                              NICE Systems
Intended status: Standards Track                             H. Lum, Ed.
Expires: November 18, 2013                                       Genesys
                                                                C. Eckel
                                                                   Cisco
                                                             A. Johnston
                                                                   Avaya
                                                               A. Hutton
                                       Siemens Enterprise Communications
                                                            May 17, 2013

                       Session Recording Protocol
                     draft-ietf-siprec-protocol-10

Abstract

   This document specifies the use of the Session Initiation Protocol
   (SIP), the Session Description Protocol (SDP), and the Real Time
   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.

Status of This Memo

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

   Internet-Drafts are working documents of the Internet Engineering
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   This Internet-Draft will expire on November 18, 2013.

Copyright Notice

   Copyright (c) 2013 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

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   This document is subject to BCP 78 and the IETF Trust's Legal
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   (http://trustee.ietf.org/license-info) in effect on the date of
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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   4
   3.  Definitions . . . . . . . . . . . . . . . . . . . . . . . . .   4
   4.  Scope . . . . . . . . . . . . . . . . . . . . . . . . . . . .   4
   5.  Overview of operations  . . . . . . . . . . . . . . . . . . .   5
     5.1.  Delivering recorded media . . . . . . . . . . . . . . . .   5
     5.2.  Delivering recording metadata . . . . . . . . . . . . . .   7
     5.3.  Receiving recording indications and providing recording
           preferences . . . . . . . . . . . . . . . . . . . . . . .   8
   6.  SIP Handling  . . . . . . . . . . . . . . . . . . . . . . . .   9
     6.1.  Procedures at the SRC . . . . . . . . . . . . . . . . . .   9
       6.1.1.  Initiating a Recording Session  . . . . . . . . . . .   9
       6.1.2.  SIP extensions for recording indication and
               preference  . . . . . . . . . . . . . . . . . . . . .  10
     6.2.  Procedures at the SRS . . . . . . . . . . . . . . . . . .  10
     6.3.  Procedures for Recording-aware User Agents  . . . . . . .  11
   7.  SDP Handling  . . . . . . . . . . . . . . . . . . . . . . . .  11
     7.1.  Procedures at the SRC . . . . . . . . . . . . . . . . . .  12
       7.1.1.  SDP handling in RS  . . . . . . . . . . . . . . . . .  12
         7.1.1.1.  Handling media stream updates . . . . . . . . . .  13
       7.1.2.  Recording indication in CS  . . . . . . . . . . . . .  13
       7.1.3.  Recording preference in CS  . . . . . . . . . . . . .  14
     7.2.  Procedures at the SRS . . . . . . . . . . . . . . . . . .  14
     7.3.  Procedures for Recording-aware User Agents  . . . . . . .  16
       7.3.1.  Recording indication  . . . . . . . . . . . . . . . .  16
       7.3.2.  Recording preference  . . . . . . . . . . . . . . . .  17
   8.  RTP Handling  . . . . . . . . . . . . . . . . . . . . . . . .  18
     8.1.  RTP Mechanisms  . . . . . . . . . . . . . . . . . . . . .  18
       8.1.1.  RTCP  . . . . . . . . . . . . . . . . . . . . . . . .  18
       8.1.2.  RTP Profile . . . . . . . . . . . . . . . . . . . . .  19
       8.1.3.  SSRC  . . . . . . . . . . . . . . . . . . . . . . . .  19
       8.1.4.  CSRC  . . . . . . . . . . . . . . . . . . . . . . . .  20
       8.1.5.  SDES  . . . . . . . . . . . . . . . . . . . . . . . .  20
         8.1.5.1.  CNAME . . . . . . . . . . . . . . . . . . . . . .  20
       8.1.6.  Keepalive . . . . . . . . . . . . . . . . . . . . . .  20
       8.1.7.  RTCP Feedback Messages  . . . . . . . . . . . . . . .  20

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         8.1.7.1.  Full Intra Request  . . . . . . . . . . . . . . .  21
         8.1.7.2.  Picture Loss Indicator  . . . . . . . . . . . . .  21
         8.1.7.3.  Temporary Maximum Media Stream Bit Rate Request .  22
       8.1.8.  Symmetric RTP/RTCP for Sending and Receiving  . . . .  22
     8.2.  Roles . . . . . . . . . . . . . . . . . . . . . . . . . .  22
       8.2.1.  SRC acting as an RTP Translator . . . . . . . . . . .  23
         8.2.1.1.  Forwarding Translator . . . . . . . . . . . . . .  23
         8.2.1.2.  Transcoding Translator  . . . . . . . . . . . . .  24
       8.2.2.  SRC acting as an RTP Mixer  . . . . . . . . . . . . .  25
       8.2.3.  SRC acting as an RTP Endpoint . . . . . . . . . . . .  25
     8.3.  RTP Session Usage by SRC  . . . . . . . . . . . . . . . .  26
       8.3.1.  SRC Using Multiple m-lines  . . . . . . . . . . . . .  26
       8.3.2.  SRC Using Mixing  . . . . . . . . . . . . . . . . . .  27
     8.4.  RTP Session Usage by SRS  . . . . . . . . . . . . . . . .  28
   9.  Metadata  . . . . . . . . . . . . . . . . . . . . . . . . . .  29
     9.1.  Procedures at the SRC . . . . . . . . . . . . . . . . . .  29
     9.2.  Procedures at the SRS . . . . . . . . . . . . . . . . . .  30
       9.2.1.  Formal Syntax . . . . . . . . . . . . . . . . . . . .  32
   10. Persistent Recording  . . . . . . . . . . . . . . . . . . . .  32
   11. IANA Considerations . . . . . . . . . . . . . . . . . . . . .  32
     11.1.  Registration of Option Tags  . . . . . . . . . . . . . .  32
       11.1.1.  siprec Option Tag  . . . . . . . . . . . . . . . . .  32
       11.1.2.  record-aware Option Tag  . . . . . . . . . . . . . .  33
     11.2.  Registration of media feature tags . . . . . . . . . . .  33
       11.2.1.  src feature tag  . . . . . . . . . . . . . . . . . .  33
       11.2.2.  srs feature tag  . . . . . . . . . . . . . . . . . .  33
     11.3.  New Content-Disposition Parameter Registrations  . . . .  34
     11.4.  Media Type Registration  . . . . . . . . . . . . . . . .  34
       11.4.1.  Registration of MIME Type application/rs-metadata  .  34
       11.4.2.  Registration of MIME Type application/rs-metadata-
                request  . . . . . . . . . . . . . . . . . . . . . .  34
     11.5.  SDP Attributes . . . . . . . . . . . . . . . . . . . . .  35
       11.5.1.  'record' SDP Attribute . . . . . . . . . . . . . . .  35
       11.5.2.  'recordpref' SDP Attribute . . . . . . . . . . . . .  35
   12. Security Considerations . . . . . . . . . . . . . . . . . . .  36
     12.1.  Authentication and Authorization . . . . . . . . . . . .  36
     12.2.  RTP handling . . . . . . . . . . . . . . . . . . . . . .  37
     12.3.  Metadata . . . . . . . . . . . . . . . . . . . . . . . .  37
     12.4.  Storage and playback . . . . . . . . . . . . . . . . . .  38
   13. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  38
   14. References  . . . . . . . . . . . . . . . . . . . . . . . . .  38
     14.1.  Normative References . . . . . . . . . . . . . . . . . .  38
     14.2.  Informative References . . . . . . . . . . . . . . . . .  39
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  40

1.  Introduction

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   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 to the architecture
   [I-D.ietf-siprec-architecture], 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.

   SIP is also used to deliver metadata to the recording device, as
   specified in [I-D.ietf-siprec-metadata].  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 requirements listed in [RFC6341].

   In addition to the Session Recording Protocol, this document
   specifies extensions for user agents that are participants in a CS to
   receive recording indications and to provide preferences for
   recording.

2.  Terminology

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

3.  Definitions

   This document refers to the core definitions provided in the
   architecture document [I-D.ietf-siprec-architecture].

   The RTP Handling section uses the definitions provided in "RTP: A
   Transport Protocol for Real-Time Application" [RFC3550].

4.  Scope

   The scope of the Session Recording Protocol includes the
   establishment of the recording sessions and the reporting of the
   metadata.  The scope also includes extensions supported by User
   Agents participating in the CS such as indication of recording.  The
   user agents need not be recording-aware in order to participate in a
   CS being recorded.

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

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   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 recording session metadata through non-SIP
      mechanism

5.  Overview of operations

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

   Section 6 describes SIP the handling in a recording session between a
   SRC and a SRS, and the procedures for recording-aware user agents
   participating in a CS.  Section 7 describes the SDP in a recording
   session, and the procedures for recording indications and recording
   preferences.  Section 8 describes the RTP handling in a recording
   session.  Section 9 describes the mechanism to deliver recording
   metadata from the SRC to the SRS.

   As mentioned in the architecture document
   [I-D.ietf-siprec-architecture], there are a number of types of call
   flows based on the location of the Session Recording Client.  The
   following sample call flows provide a quick overview of the
   operations between the SRC and the SRS.

5.1.  Delivering recorded media

   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 user agents.  When the SRC is aware that it should
   be recording the conversation, the SRC can cause the B2BUA to bridge
   the media between UA(A) and UA(B).  The SRC then establishes the
   Recording Session with the SRS and sends replicated media towards the
   SRS.

   An endpoint may also have SRC functionality, where the endpoint
   itself establishes the Recording Session to the SRS.  Since the

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   endpoint has access to the media in the Communication Session, the
   endpoint can send replicated media towards the SRS.

   The following is a sample call flow that shows the SRC establishing a
   recording session towards the SRS.  The call flow is essentially
   identical when the SRC is a B2BUA or as the endpoint itself.  Note
   that the SRC can choose when to establish the Recording Session
   independent of the Communication Session, even though the following
   call flow suggests that the SRC is establishing the Recording Session
   (message #5) after the Communication Session is established.

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

                    Figure 1: Basic recording call flow

   The above call flow 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

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   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 recording session to record multiple
   communication sessions.  Every time the SRC wants to record a new
   call, the SRC updates the recording session with a new SDP offer to
   add new recorded streams to the recording session, and
   correspondingly also update the metadata for the new call.

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

5.2.  Delivering recording metadata

   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 recording session.
   Subsequent metadata updates can be represented as a stream of events
   in UPDATE or reINVITE 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 anytime.

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

   The SRS has the ability to send a request to the SRC to request 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 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 SRC and SRS to
   synchronize the states with a new metadata snapshot so that further
   metadata incremental 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 sent by the SRS
   in the recording session.

     SRC                                                   SRS
      |                                                     |
      |(1) INVITE (metadata snapshot)                       |
      |---------------------------------------------------->|
      |                                           (2)200 OK |
      |<----------------------------------------------------|

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

               Figure 2: Delivering metadata via SIP UPDATE

5.3.  Receiving recording indications and providing recording
      preferences

   The SRC is responsible to provide recording indications to the
   participants in the CS.  A recording-aware UA supports receiving
   recording indications via the SDP attribute a=record, and it can
   specify a recording preference in the CS by including the SDP
   attribute a=recordpref.  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 the recording preference.

   To illustrate how the attributes are used, if a 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

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   provide any recording preference, the SDP answer does not contain
   a=record nor a=recordpref.

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

          Figure 3: 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 reINVITE with the a=recordpref attribute.  It is up to the
   SRC to honor the preference, and in this case SRC decides to stop the
   recording and updates the recording indication in the SDP answer.

6.  SIP Handling

6.1.  Procedures at the SRC

6.1.1.  Initiating a Recording Session

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

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   The SRC can initiate a recording session 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 [RFC3840], for all
   recording sessions.  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 a
   Recording Session.  In addition, when an SRC sends a REGISTER request
   to a registrar, the SRC MUST include the '+sip.src' feature tag to
   indicate the that it is a SRC.

   Since SIP Caller Preferences extensions are optional to implement for
   routing proxies, there is no guarantee that a recording session will
   be routed to an SRC or SRS.  A new options tag is introduced:
   "siprec".  As per [RFC3261], only an SRC or an SRS can accept this
   option tag in a recording session.  An SRC MUST include the "siprec"
   option tag in the Require header when initiating a Recording Session
   so that UA's which do not support the session recording protocol
   extensions will simply reject the INVITE request with a 420 Bad
   Extension.

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

6.1.2.  SIP extensions for recording indication and preference

   For the communication session, the SRC MUST provide recording
   indication 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 the SDP Handling section.
   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 inband, if the
   SRC is required to do so (e.g.  based on policies).

   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 a SRC is in the CS, and can set the
   recording preference for the CS with the new SDP a=recordpref
   attribute described in the SDP Handling section below.

6.2.  Procedures at the SRS

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

   The SRS can initiate a recording session 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 [RFC3840], for all recording sessions.  An SRC uses the
   presence of this feature tag in dialog creating and modifying
   requests and responses to confirm that the dialog being created is
   for the purpose of a Recording Session (REQ-30).  In addition, when
   an SRS sends a REGISTER request to a registrar, the SRS MUST include
   the '+sip.srs' feature tag to indicate that it is a SRS.

   An SRS MUST include the "siprec" option tag in the Require header as
   per [RFC3261] when initiating a Recording Session so that UA's which
   do not support the session recording protocol extensions will simply
   reject the INVITE request with a 420 Bad Extension.

6.3.  Procedures for Recording-aware User Agents

   A recording-aware user agent is a participant in the CS that supports
   the SIP and SDP extensions for receiving recording indication and for
   requesting recording preferences for the call.  A recording-aware UA
   MUST indicate that it can accept reporting of recording indication
   provided by the SRC with a new option tag "record-aware" when
   initiating or establishing a CS, meaning including the "record-aware"
   tag in the Supported header in the initial INVITE request or
   response.

   A recording-aware UA MUST be prepared to provide recording indication
   to the end user through an appropriate user interface an indication
   whether recording is on, off, or paused for each medium.  Some user
   agents that are automatons (e.g.  IVR, media server, PSTN gateway)
   may not have a user interface to render recording indication.  When
   such user agent indicates recording awareness, the UA SHOULD render
   recording indication through other means, such as passing an inband
   tone on the PSTN gateway, putting the recording indication in a log
   file, or raising an application event in a VoiceXML dialog.  These
   user agents MAY also choose not to indicate recording awareness,
   thereby relying on whatever mechanism an SRC chooses to indicate
   recording, such as playing a tone inband.

7.  SDP Handling

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7.1.  Procedures at the SRC

   The SRC and SRS follows the SDP offer/answer model in [RFC3264].  The
   procedures for SRC and SRS describe the conventions used in a
   recording session.

7.1.1.  SDP handling in RS

   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 [RFC3264].

   The SRC sends recorded streams of participants to the SRS, and the
   SRC MUST provide a label attribute (a=label), as per [RFC4574], 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 [I-D.ietf-siprec-metadata].  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 contributes
   to each recorded stream.

   The following is an example SDP offer from SRC with both audio and
   video recorded streams.  Note that the following example contains
   unfolded lines longer than 72 characters.  These are captured between
   <allOneLine> 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
       <allOneLine>
       a=fmtp:98 profile-level-id=42A01E;
                 sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==
       </allOneLine>
       a=sendonly
       a=label:2
       m=audio 12242 RTP/AVP 0 4 8

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       a=sendonly
       a=label:3
       m=video 22458 RTP/AVP 98
       a=rtpmap:98 H264/90000
       <allOneLine>
       a=fmtp:98 profile-level-id=42A01E;
                 sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==
       </allOneLine>
       a=sendonly
       a=label:4

     Figure 4: Sample SDP offer from SRC with audio and video streams

7.1.1.1.  Handling media stream updates

   Over the lifetime of a recording session, the SRC can add and remove
   recorded streams from the recording session for various reasons.  For
   example, when a CS stream is added or removed from the CS, or when a
   CS is created or terminated if a recording session handles multiple
   CSes.  To remove a recorded stream from the recording session, 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 [RFC3264].  To
   add a recorded stream to the recording session, the SRC sends a new
   SDP offer by adding a new media stream description or by reusing an
   old media stream which had been previously disabled, according to the
   procedures in [RFC3264].

   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 [RFC3264].  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
   itself may change the media stream direction by updating the SDP, for
   example, by setting a=inactive for SDP hold.  Media stream direction
   changes in CS are conveyed in the metadata by the SRC.  The SRC MUST
   NOT modify the media stream with a=inactive for SDP hold on the CS
   since this operation is reserved for pausing the RS media, however,
   an SRC can have a local policy to pause the RS media when the CS is
   placed on hold.

7.1.2.  Recording indication in CS

   While there are existing mechanisms for providing an indication that
   a CS is being recorded, these mechanisms are usually delivered on the

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   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
   CS.

   The 'record' SDP attribute appears at the media level or session
   level in either 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 the 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, then
   the SRC MUST initiate a reINVITE or UPDATE to update the SDP a=record
   attribute.

   The following is the ABNF of the 'record' attribute:

      attribute /= record-attr

      ; attribute defined in RFC 4566

      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.

7.1.3.  Recording preference in CS

   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.  Whether or not the SRC honors the recording
   preference, the SRC MUST update the a=record attribute to indicate
   the current state of the recording (on/off/paused).

7.2.  Procedures at the SRS

   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 [RFC3264].  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 a=inactive
   attribute, according to the procedures of [RFC3264].  When the SRS is

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   ready to receive recorded streams, the SRS sends a new SDP offer and
   sets the media streams with a=recvonly attribute.

   The following is an example of SDP answer from SRS for the SDP offer
   from the above sample.  Note that the following example contain
   unfolded lines longer than 72 characters.  These are captured between
   <allOneLine> 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
       <allOneLine>
       a=fmtp:98 profile-level-id=42A01E;
                 sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==
       </allOneLine>
       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
       <allOneLine>
       a=fmtp:98 profile-level-id=42A01E;
                 sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==
       </allOneLine>
       a=recvonly
       a=label:4

     Figure 5: Sample SDP answer from SRS with audio and video streams

   Over the lifetime of a recording session, the SRS can remove recorded
   streams from the recording session for various reasons.  To remove a
   recorded stream from the recording session, 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 [RFC3264].

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

   The following sequence diagram shows an example where the SRS is
   initially not ready to receive recorded streams, and later updates
   the recording session when the SRS is ready to record.

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

             Figure 6: SRS responding to offer with a=inactive

7.3.  Procedures for Recording-aware User Agents

7.3.1.  Recording indication

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

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   If a call is traversed through one or more SIP B2BUA, and it happens
   that there are more than one SRC in the call path, the recording
   indication attribute does not provide any hint as to which SRC is
   performing the recording, meaning the endpoint only knows that the
   call is being recorded.  This attribute is also not used as an
   indication to negotiate which SRC in the call path will perform
   recording and is not used as a request to start/stop recording if
   there are multiple SRCs in the call path.

7.3.2.  Recording preference

   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 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 the above section.

   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 stream in the SDP.  When the attribute is
   applied at the media level, the recording preference applies to the
   media stream only, and that overrides the recording preference if
   also set at the session level.  The user agent can change the
   recording preference by changing the a=recordpref attribute in
   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 RFC 4566

      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.

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

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

   nopreference  To indicate that the UA has no preference on recording.

8.  RTP Handling

   This section provides recommendations and guidelines for RTP and RTCP
   in the context of SIPREC.  In order to communicate most effectively,
   the Session Recording Client (SRC), the Session Recording Server
   (SRS), and any Recording aware User Agents (UAs) SHOULD utilize the
   mechanisms provided by RTP in a well-defined and predicable manner.
   It is the goal of this document to make the reader aware of these
   mechanisms and provide recommendations and guidelines.

8.1.  RTP Mechanisms

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

8.1.1.  RTCP

   The RTP data transport is augmented by a control protocol (RTCP) to
   allow monitoring of the data delivery.  RTCP, as defined in
   [RFC3550], 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
   [RFC3550], 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) Carries a persistent transport-level identifier for an RTP source
   called the canonical name or CNAME

   The SSRC 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 multiple data streams from a given participant in a set of

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   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.

8.1.2.  RTP Profile

   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)", [RFC5124] when using
   encrypted RTP streams, and "Extended RTP Profile for Real-time
   Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)",
   [RFC4585] when using non encrypted media streams.  However, as this
   is not a requirement, some implementations may use "The Secure Real-
   time Transport Protocol (SRTP)", [RFC3711] and "RTP Profile for Audio
   and Video Conferences with Minimal Control", AVP [RFC3551].
   Therefore, it is RECOMMENDED that the SRC, SRS, and Recording aware
   UAs not rely entirely on SAVPF or AVPF for core functionality that
   may be at least partially achievable using SAVP and 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, 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.

8.1.3.  SSRC

   The synchronization source (SSRC), as defined in [RFC3550] 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
   [RFC3550].

   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 of
   the change.  The CNAME values carried in RTCP facilitate this
   identification.

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8.1.4.  CSRC

   The contributing source (CSRC), as defined in [RFC3550], 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 a SRC or
   Recording aware UA, when acting a mixer, sets the CSRC list
   accordingly, and that the SRC and SRS interpret the CSRC list
   appropriately when received.

8.1.5.  SDES

   The Source Description (SDES), as defined in [RFC3550], contains an
   SSRC/CSRC identifier followed by a list of zero or more items, which
   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 contributing source from which it is receiving SDES
   information, or multiple complete SDES packets if there are more than
   31 such sources.

8.1.5.1.  CNAME

   The Canonical End-Point Identifier (CNAME), as defined in [RFC3550],
   provides the binding from the SSRC identifier to an identifier for
   the source (sender or receiver) that remains constant.  It is
   important 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)" [RFC6222].

8.1.6.  Keepalive

   It is anticipated that media streams in SIPREC may exist in an
   inactive state for extended periods of times 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 keep-alive
   procedure recommended in "Application Mechanism for Keeping Alive the
   NAT Mappings Associated to RTP/RTP Control Protocol (RTCP) Flows"
   [RFC6263] for all RTP media streams.

8.1.7.  RTCP Feedback Messages

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   "Codec Control Messages in the RTP Audio-Visual Profile with Feedback
   (AVPF)" [RFC5104] specifies extensions to the messages defined in
   AVPF [RFC4585].  Support for and proper usage of these messages is
   important to SRC, SRS, and Recording aware UA implementations.  Note
   that these messages are applicable only when using the AVFP or SAVPF
   RTP profiles

8.1.7.1.  Full Intra Request

   A Full Intra Request (FIR) Command, when received by the designated
   media sender, requires that the media sender sends 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 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.

8.1.7.1.1.  SIP INFO for FIR

   "XML Schema for Media Control" [RFC5168] defines an Extensible Markup
   Language (XML) Schema for video fast update.  Implementations are
   discouraged from using the method described except for backward
   compatibility purposes.  Implementations SHOULD use FIR messages
   instead.

8.1.7.2.  Picture Loss Indicator

   Picture Loss Indication (PLI), as defined in [RFC4585], informs the
   encoder of the loss of an undefined amount of coded video data
   belonging to one or more pictures.  Using the FIR command to recover
   from errors is explicitly disallowed, and instead the PLI message
   SHOULD be used.  FIR SHOULD be used only in situations where not
   sending a decoder refresh point would render the video unusable for
   the users.  Examples where sending FIR is appropriate include a
   multipoint conference when a new user joins the conference and no
   regular decoder refresh point interval is established, and a video
   switching MCU that changes streams.

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8.1.7.3.  Temporary Maximum Media Stream Bit Rate Request

   A receiver, translator, or mixer uses the Temporary Maximum Media
   Stream Bit Rate Request (TMMBR) 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.

8.1.7.3.1.  Renegotiation of SDP bandwidth attribute

   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.

8.1.8.  Symmetric RTP/RTCP for Sending and Receiving

   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 as those signaled
   for receiving packets.  When the transport address used to send and
   receive RTP is the same, it is termed "symmetric RTP" [RFC4961].
   Likewise, when the transport address used to send and receive RTCP is
   the same, it is termed "symmetric RTCP" [RFC4961].

   When sending RTP, it is REQUIRED to use symmetric RTP.  When sending
   RTCP, it is REQUIRED to use symmetric RTCP.  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 [RFC5761].

8.2.  Roles

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

          (Recording Session)   +---------+
        +------------SIP------->|         |

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        |  +------RTP/RTCP----->|   SRS   |
        |  |    +-- Metadata -->|         |
        |  |    |               +---------+
        v  v    |
       +---------+
       |   SRC   |
       |---------| (Communication Session) +---------+
       |         |<----------SIP---------->|         |
       |  UA-A   |                         |  UA-B   |
       |         |<-------RTP/RTCP-------->|         |
       +---------+                         +---------+

                            Figure 7: UA as SRC

                               (Recording Session)   +---------+
                             +------------SIP------->|         |
                             |  +------RTP/RTCP----->|   SRS   |
                             |  |    +-- Metadata -->|         |
                             |  |    |               +---------+
                             v  v    |
                            +---------+
                            |   SRC   |
   +---------+              |---------|              +---------+
   |         |<----SIP----->|         |<----SIP----->|         |
   |  UA-A   |              |  B2BUA  |              |  UA-B   |
   |         |<--RTP/RTCP-->|         |<--RTP/RTCP-->|         |
   +---------+              +---------+              +---------+
         |_______________________________________________|
                      (Communication Session)

                          Figure 8: 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.

8.2.1.  SRC acting as an RTP Translator

   The SRC may act as a translator, as defined in [RFC3550].  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.

8.2.1.1.  Forwarding Translator

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   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 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 not of relevance.

   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.
   Section 12 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.

8.2.1.2.  Transcoding Translator

   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 and what
   the SRC sends.  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.

   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 account of 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

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   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 not of relevance.

   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.
   Section 12 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.

8.2.2.  SRC acting as an RTP Mixer

   In the case of the SRC acting as a RTP mixer, as defined in
   [RFC3550], the SRC combines RTP streams from different UA and sends
   them towards the SRS using its own SSRC.  The SSRCs from the
   contributing UA SHOULD be conveyed as CSRCs 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 mixer, but there are requirements for forwarding RTCP
   Source Description (SDES) packets.  The SRC generates its own RTCP
   Sender 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 SRC for the CS.  Section 12
   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.

8.2.3.  SRC acting as an RTP Endpoint

   The case of the SRC acting as an RTP endpoint, as defined in
   [RFC3550], 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.

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   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.  Section 12
   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.

8.3.  RTP Session Usage by SRC

   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.

8.3.1.  SRC Using Multiple m-lines

   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
   [RFC3264].  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.

   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 to, the SRC may act as an RTP
   translator or as an RTP endpoint.

   The following figure 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.

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                                                    +---------+
                            +------------SSRC Aa--->|         |
                            |  + --------SSRC Av--->|         |
                            |  |  +------SSRC Ba--->|   SRS   |
                            |  |  |  +---SSRC Bv--->|         |
                            |  |  |  |              +---------+
                            |  |  |  |
                            |  |  |  |
   +---------+             +----------+             +---------+
   |         |---SSRC Aa-->|   SRC    |<--SSRC Ba---|         |
   |  UA-A   |             |(CNAME-A, |             |  UA-B   |
   |(CNAME-A)|---SSRC Av-->| CNAME-B) |<--SSRC Bv---|(CNAME-B)|
   +---------+             +----------+             +---------+

                   Figure 9: SRC Using Multiple m-lines

8.3.2.  SRC Using Mixing

   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 CSRCs
   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 the zero port, per
   [RFC3264].  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 CNAME/
   SSRC within the RS.  The SRC MUST avoid SSRC collisions, rewriting
   SSRCs if necessary when used as CSRCs in the RS.  In doing to, the
   SRC acts as an RTP mixer.

   In the event 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.

   The following figure 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--->|         |

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                              |                     |         |
                              |       SSRC Sa       |   SRS   |
                              |   +---CSRC Av,Bv--->|         |
                              |   |                 +---------+
                              |   |
                           +----------+
   +---------+             |   SRC    |             +---------+
   |         |---SSRC Aa-->|(CNAME-S, |<--SSRC Ba---|         |
   |  UA-A   |             | CNAME-A, |             |  UA-B   |
   |(CNAME-A)|---SSRC Aa-->| CNAME-B) |<--SSRC Bv---|(CNAME-B)|
   +---------+             +----------+             +---------+

                        Figure 10: SRC Using Mixing

8.4.  RTP Session Usage by SRS

   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 an 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.

   Note that 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
   other modes 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 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 have to at
   least support the ones above such that they interoperate in the
   common base mode for basic interoperability.

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9.  Metadata

9.1.  Procedures at the SRC

   The SRC MUST deliver metadata to the SRS in a recording session; the
   timing of which SRC sends the metadata depends on when the metadata
   becomes available.  Metadata SHOULD be provided by the SRC in the
   initial INVITE request when establishing the recording session, and
   subsequent metadata updates can be provided by the SRC in reINVITE
   and UPDATE requests ([RFC3311]) and responses in the recording
   session.  There are cases that metadata is not available in the
   initial INVITE request sent by the SRC, for example, when a recording
   session is established in the absence of a communication session, and
   the SRC would update the recording session with metadata whenever
   metadata becomes available.

   Certain metadata attributes are contained in the SDP, and others are
   contained in a new content type "application/rs-metadata".  The
   format of the metadata is described as part of the mechanism in
   [I-D.ietf-siprec-metadata].  A new "disposition-type" of Content-
   Disposition is defined for the purpose of carrying metadata and the
   value is "recording-session".  The "recording-session" value
   indicates that the "application/rs-metadata" content contains
   metadata to be handled by the SRS, and the disposition can be carried
   in either INVITE or UPDATE requests or responses sent by the SRC.

   Metadata sent by the SRC can be categorized as either a full metadata
   snapshot or partial update.  A full metadata snapshot describes all
   the recorded streams and all metadata associated with the recording
   session.  When the SRC sends a full metadata snapshot, the SRC MUST
   send an INVITE or an UPDATE request ([RFC3311]) with an SDP offer and
   the "recording-session" disposition.  A partial update represents an
   incremental update since the last metadata update sent by the SRC.  A
   partial update sent by the SRC can be an INVITE request or response
   with an SDP offer, or an INVITE/UPDATE request or response containing
   a "recording-session" disposition, or an INVITE request containing
   both an SDP offer and the "recording-session" disposition.

   The following is 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: <sip:2000@example.com>;tag=35e195d2-947d-4585-946f-09839247
       To: <sip:recorder@example.com>
       Call-ID: d253c800-b0d1ea39-4a7dd-3f0e20a
       CSeq: 101 INVITE

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       Max-Forwards: 70
       Require: siprec
       Accept: application/sdp, application/rs-metadata,
         application/rs-metadata-request
       Contact: <sip:2000@src.example.com>;+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 11: Sample INVITE request for the recording session

9.2.  Procedures at the SRS

   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 for 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
   error on the SRC side, and it is likely 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
   recording session when a syntax error or semantic error is detected
   in the metadata.

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   When the SRS explicitly requests for a full metadata snapshot, the
   SRS MUST send an UPDATE request without an SDP offer.  A metadata
   snapshot request contains a content with the content disposition type
   "recording-session".  Note that the SRS MAY generate an INVITE
   request without an SDP offer but this MUST NOT include a metadata
   snapshot request.  The format of the content is "application/rs-
   metadata-request", and the body format is chosen to be a simple text-
   based format.  The following shows an example:

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

       SRS internal error

                        Figure 12: Metadata Request

   The SRS MAY include the reason why a metadata snapshot request is
   being made to the SRC in the reason line.  This reason line is free
   form text, mainly designed for logging purposes on the SRC side.  The
   processing of the content by the SRC is entirely optional since the
   content is for logging only, and the snapshot request itself is
   indicated by the use of the application/rs-metadata-request content
   type.

   When the SRC receives the request for a metadata snapshot, the SRC
   MUST provide a full metadata snapshot in a separate INVITE or UPDATE
   transaction, along with an SDP offer.  All subsequent metadata
   updates sent by the SRC MUST be based on the new metadata snapshot.

   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

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   endpoint in the same CS act as an SRC to record the call and a common
   ID refers to the same CS.  The SRS MUST NOT consider this an error.

9.2.1.  Formal Syntax

   The formal syntax for the application/rs-metadata-request MIME is
   described below using the augmented Backus-Naur Form (BNF) as
   described in [RFC5234].

   snapshot-request = srs-reason-line CRLF

   srs-reason-line = [TEXT-UTF8-TRIM]

10.  Persistent Recording

   Persistent recording is a specific use case outlined in REQ-005 or
   Use Case 4 in [RFC6341], where a recording session can be established
   in the absence of a communication session.  The SRC continuously
   records media in a recording session to the SRS even in the absence
   of a CS for all user agents 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 new SDP offer when a new communication session is
   created and also does not impact the timing of the CS.  The SRC only
   needs to update the metadata when new communication sessions are
   created.

   When there is no communication sessions running on the devices with
   persistent recording, there is no recorded media to stream from the
   SRC to the SRS.  In certain environments where Network Address
   Translator (NAT) is used, typically a minimum of flow activity is
   required to maintain the NAT binding for each port opened.  Agents
   that support Interactive Connectivity Establishment (ICE) solves 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 [RFC6263] to
   maintain the NAT bindings.

11.  IANA Considerations

11.1.  Registration of Option Tags

   This specification registers two option tags.  The required
   information for this registration, as specified in [RFC3261], is as
   follows.

11.1.1.  siprec Option Tag

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      Name: siprec

      Description: This option tag is for identifying the SIP session
      for the purpose of recording session only.  This is typically not
      used in a Supported header.  When present in a Require header in a
      request, it indicates that the UAS MUST be either a SRC or SRS
      capable of handling the contexts of a recording session.

11.1.2.  record-aware Option Tag

      Name: record-aware

      Description: This option tag is to indicate the ability for the
      user agent 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.

11.2.  Registration of media feature tags

   This document registers two new media feature tags in the SIP tree
   per the process defined in [RFC2506] and [RFC3840]

11.2.1.  src feature tag

      Media feature tag name: sip.src

      ASN.1 Identifier: 25

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

      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 a Recording Session.

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

      Security Considerations: Security considerations for this media
      feature tag are discussed in Section 11.1 of RFC 3840.

11.2.2.  srs feature tag

      Media feature tag name: sip.srs

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      ASN.1 Identifier: 26

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

      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 a Recording Session.

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

      Security Considerations: Security considerations for this media
      feature tag are discussed in Section 11.1 of RFC 3840.

11.3.  New Content-Disposition Parameter Registrations

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

   recording-session the body describes the metadata information about
   the recording session

11.4.  Media Type Registration

11.4.1.  Registration of MIME Type application/rs-metadata

   This document registers the application/rs-metadata MIME media type
   in order to describe the recording session metadata.  This media type
   is defined by the following information:

   Media type name: application

   Media subtype name: rs-metadata

   Required parameters: none

   Options parameters: none

11.4.2.  Registration of MIME Type application/rs-metadata-request

   This document registers the application/rs-metadata-request MIME
   media type in order to describe a recording session metadata snapshot
   request.  This media type is defined by the following information:

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   Media type name: application

   Media subtype name: rs-metadata-request

   Required parameters: none

   Options parameters: none

11.5.  SDP Attributes

   This document registers the following new SDP attributes.

11.5.1.  'record' SDP Attribute

   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 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

11.5.2.  'recordpref' SDP Attribute

   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 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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12.  Security Considerations

   The recording session is fundamentally a standard SIP dialog
   [RFC3261], therefore, the recording session 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 [RFC6341] outlines the general security
   considerations, and this document describes specific security
   recommendations.

   The SRC and SRS MUST support SIP with TLS and MAY support SIPS with
   TLS as per [RFC5630].  The Recording Session SHOULD be at least as
   secure as the Communication Session, meaning 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 should not downgrade the level of security in the RS to
   SIP or plain RTP since doing so will mean an automatic security
   downgrade for the CS.  In deployments where the SRC and the SRS are
   in the same administrative domain and the same physical switch that
   prevents outside user access, some SRCs may choose to lower the level
   of security when establishing a recording session.  While physically
   securing the SRC and SRS may prevent an outside attacker from
   accessing important call recordings, this still does not prevent an
   inside attacker from accessing the internal network to gain access to
   the call recordings.

12.1.  Authentication and Authorization

   At the transport level, the recording session uses TLS authentication
   to validate the authenticity of the SRC and SRS.  The SRC and SRS
   MUST implement TLS mutual authentication for establishing the
   recording session.  Whether the SRC/SRS chooses to use TLS mutual
   authentication is a deployment decision.  In deployments where the
   SRC and the SRS are in the same administrative domain, the SRC and
   SRS may choose not to authenticate each other, or to have the SRC
   authenticate the SRS only, as there is an inherent trust relation
   between the SRC and the SRS when they are hosted in the same
   administrative domain.  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 a
   compromised SRS and that a sensitive call recording will be obtained
   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.

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   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
   utilized.  Additionally, an SRC or SRS may use other existing SIP
   mechanisms available, including but not limited to, Digest
   Authentication [RFC3261], Asserted Identity [RFC3325], and Connected
   Identity [RFC4916].

   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.

12.2.  RTP handling

   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 [RFC3711] and RTP/SAVPF [RFC5124].
   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 (SDES) key negotiation mechanism [RFC4568].
   For cases in which DTLS-SRTP is used to encrypt a CS media stream, an
   SRC may use SRTP Encrypted Key Transport (EKT)
   [I-D.ietf-avt-srtp-ekt] in order to use SRTP-SDES in the RS without
   needing to re-encrypt the media.

   When RTP/SAVP or RTP/SAVPF is used, an SRC can choose to use the same
   or different keys in the RS than the ones 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
   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 CS and RS.

   SRCs that decrypt an encrypted CS media stream and re-encrypt it when
   sending it to the SRS MUST use a different key for the RS media
   stream 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 are not authorized to access.

12.3.  Metadata

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   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.

12.4.  Storage and playback

   While storage and playback of the call recording is 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 communication session.  It
   would defeat the purpose of securing both the communication session
   and the recording session 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.

13.  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.

14.  References

14.1.  Normative References

   [I-D.ietf-siprec-metadata]
              R, R., Ravindran, P., and P. Kyzivat, "Session Initiation
              Protocol (SIP) Recording Metadata", draft-ietf-siprec-
              metadata-11 (work in progress), January 2013.

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

   [RFC2506]  Holtman, K., Mutz, A., and T. Hardie, "Media Feature Tag
              Registration Procedure", BCP 31, RFC 2506, March 1999.

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              June 2002.

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

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   [RFC3840]  Rosenberg, J., Schulzrinne, H., and P. Kyzivat,
              "Indicating User Agent Capabilities in the Session
              Initiation Protocol (SIP)", RFC 3840, August 2004.

   [RFC4574]  Levin, O. and G. Camarillo, "The Session Description
              Protocol (SDP) Label Attribute", RFC 4574, August 2006.

   [RFC5234]  Crocker, D. and P. Overell, "Augmented BNF for Syntax
              Specifications: ABNF", STD 68, RFC 5234, January 2008.

14.2.  Informative References

   [I-D.ietf-avt-srtp-ekt]
              Wing, D., McGrew, D., and K. Fischer, "Encrypted Key
              Transport for Secure RTP", draft-ietf-avt-srtp-ekt-03
              (work in progress), October 2011.

   [I-D.ietf-siprec-architecture]
              Hutton, A., Portman, L., Jain, R., and K. Rehor, "An
              Architecture for Media Recording using the Session
              Initiation Protocol", draft-ietf-siprec-architecture-07
              (work in progress), November 2012.

   [RFC3311]  Rosenberg, J., "The Session Initiation Protocol (SIP)
              UPDATE Method", RFC 3311, October 2002.

   [RFC3325]  Jennings, C., Peterson, J., and M. Watson, "Private
              Extensions to the Session Initiation Protocol (SIP) for
              Asserted Identity within Trusted Networks", RFC 3325,
              November 2002.

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

   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
              Video Conferences with Minimal Control", STD 65, RFC 3551,
              July 2003.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, March 2004.

   [RFC4568]  Andreasen, F., Baugher, M., and D. Wing, "Session
              Description Protocol (SDP) Security Descriptions for Media
              Streams", RFC 4568, July 2006.

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   [RFC4585]  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)", RFC 4585, July
              2006.

   [RFC4916]  Elwell, J., "Connected Identity in the Session Initiation
              Protocol (SIP)", RFC 4916, June 2007.

   [RFC4961]  Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
              BCP 131, RFC 4961, July 2007.

   [RFC5104]  Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
              "Codec Control Messages in the RTP Audio-Visual Profile
              with Feedback (AVPF)", RFC 5104, February 2008.

   [RFC5124]  Ott, J. and E. Carrara, "Extended Secure RTP Profile for
              Real-time Transport Control Protocol (RTCP)-Based Feedback
              (RTP/SAVPF)", RFC 5124, February 2008.

   [RFC5168]  Levin, O., Even, R., and P. Hagendorf, "XML Schema for
              Media Control", RFC 5168, March 2008.

   [RFC5630]  Audet, F., "The Use of the SIPS URI Scheme in the Session
              Initiation Protocol (SIP)", RFC 5630, October 2009.

   [RFC5761]  Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
              Control Packets on a Single Port", RFC 5761, April 2010.

   [RFC6222]  Begen, A., Perkins, C., and D. Wing, "Guidelines for
              Choosing RTP Control Protocol (RTCP) Canonical Names
              (CNAMEs)", RFC 6222, April 2011.

   [RFC6263]  Marjou, X. and A. Sollaud, "Application Mechanism for
              Keeping Alive the NAT Mappings Associated with RTP / RTP
              Control Protocol (RTCP) Flows", RFC 6263, June 2011.

   [RFC6341]  Rehor, K., Portman, L., Hutton, A., and R. Jain, "Use
              Cases and Requirements for SIP-Based Media Recording
              (SIPREC)", RFC 6341, August 2011.

Authors' Addresses

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   Leon Portman
   NICE Systems
   8 Hapnina
   Ra'anana  43017
   Israel

   Email: leon.portman@nice.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
   Avaya
   St. Louis, MO  63124

   Email: alan.b.johnston@gmail.com

   Andrew Hutton
   Siemens Enterprise Communications
   Brickhill Street
   Milton Keynes  MK15 0DJ
   United Kingdom

   Email: andrew.hutton@siemens-enterprise.com

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