SIPREC                                                        L. Portman
Internet-Draft                                              NICE Systems
Intended status: Standards Track                             H. Lum, Ed.
Expires: January 14, April 5, 2013                                           Genesys
                                                                C. Eckel
                                                                   Cisco
                                                             A. Johnston
                                                                   Avaya
                                                               A. Hutton
                                                      Siemens Enterprise
                                                          Communications
                                                           July 13,
                                                         October 2, 2012

                       Session Recording Protocol
                     draft-ietf-siprec-protocol-06
                     draft-ietf-siprec-protocol-07

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

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   This Internet-Draft will expire on January 14, April 5, 2013.

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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  4
   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.  Initiating a Recording Session  SIP Handling . . . . . . . . . . . . . . . . . . . . . . . . .  9
     6.1.  Procedures at the SRC  . . . . . . . . . . . . . . . . . . 10
       6.1.1.  Initiating a Recording Session . . . . . . . . . . . . 10
       6.1.2.  SIP extensions for recording indication and
               preference . . . . . . . . . . . . . . . . . . . . . . 10
     6.2.  Procedures at the SRS  . . . . . . . . . . . . . . . . . . 10 11
     6.3.  Procedures for Recording-aware User Agents . . . . . . . . 11
   7.  SDP Handling . . . . . . . . . . . . . . . . . . . . . . . . . 11 12
     7.1.  Procedures at the SRC  . . . . . . . . . . . . . . . . . . 11 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 . . . 12 . . . . . . . . . . . 14
       7.1.3.  Recording preference in CS . . . . . . . . . . . . . . 15
     7.2.  Procedures at the SRS  . . . . . . . . . . . . . . . . . . 13 15
     7.3.  Procedures for Recording-aware User Agents . . . . . . . . 17
       7.3.1.  Recording indication . . . . . . . . . . . . . . . . . 17
       7.3.2.  Recording preference . . . . . . . . . . . . . . . . . 18
   8.  RTP Handling . . . . . . . . . . . . . . . . . . . . . . . . . 15 19
     8.1.  RTP Mechanisms . . . . . . . . . . . . . . . . . . . . . . 15 19
       8.1.1.  RTCP . . . . . . . . . . . . . . . . . . . . . . . . . 16 19
       8.1.2.  RTP Profile  . . . . . . . . . . . . . . . . . . . . . 16 19
       8.1.3.  SSRC . . . . . . . . . . . . . . . . . . . . . . . . . 17 20
       8.1.4.  CSRC . . . . . . . . . . . . . . . . . . . . . . . . . 17 20
       8.1.5.  SDES . . . . . . . . . . . . . . . . . . . . . . . . . 17 21
         8.1.5.1.  CNAME  . . . . . . . . . . . . . . . . . . . . . . 18 21
       8.1.6.  Keepalive  . . . . . . . . . . . . . . . . . . . . . . 18 21
       8.1.7.  RTCP Feedback Messages . . . . . . . . . . . . . . . . 18 21
         8.1.7.1.  Full Intra Request . . . . . . . . . . . . . . . . 18 22
         8.1.7.2.  Picture Loss Indicator . . . . . . . . . . . . . . 19 22
         8.1.7.3.  Temporary Maximum Media Stream Bit Rate Request  . 19 22
       8.1.8.  Symmetric RTP/RTCP for Sending and Receiving . . . . . 19 23
     8.2.  Roles  . . . . . . . . . . . . . . . . . . . . . . . . . . 20 23
       8.2.1.  SRC acting as an RTP Translator  . . . . . . . . . . . 21 24
         8.2.1.1.  Forwarding Translator  . . . . . . . . . . . . . . 21 25
         8.2.1.2.  Transcoding Translator . . . . . . . . . . . . . . 22 25
       8.2.2.  SRC acting as an RTP Mixer . . . . . . . . . . . . . . 23 26
       8.2.3.  SRC acting as an RTP Endpoint  . . . . . . . . . . . . 23 26
     8.3.  RTP Session Usage by SRC . . . . . . . . . . . . . . . . . 23 27
       8.3.1.  SRC Using Multiple m-lines . . . . . . . . . . . . . . 24 27
       8.3.2.  SRC Using SSRC Multiplexing  . . . . . . . . . . . . . 25 28
       8.3.3.  SRC Using Mixing . . . . . . . . . . . . . . . . . . . 26 29
   9.  Metadata . . . . . . . . . . . . . . . . . . . . . . . . . . . 27 30
     9.1.  Procedures at the SRC  . . . . . . . . . . . . . . . . . . 27 30
     9.2.  Procedures at the SRS  . . . . . . . . . . . . . . . . . . 29 32
       9.2.1.  Formal Syntax  . . . . . . . . . . . . . . . . . . . . 30 34
   10. Persistent Recording . . . . . . . . . . . . . . . . . . . . . 30 34
   11. Extensions for Recording-aware User Agents . . . . . . . . . . 31
     11.1. Procedures at the record-aware user agent  . . . . . . . . 31
       11.1.1. Recording preference . . . . . . . . . . . . . . . . . 31
     11.2. Procedures at the SRC  . . . . . . . . . . . . . . . . . . 32
       11.2.1. Recording indication . . . . . . . . . . . . . . . . . 32
       11.2.2. Recording preference . . . . . . . . . . . . . . . . . 33
   12. IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 33
     12.1. 34
     11.1. Registration of Option Tags  . . . . . . . . . . . . . . . 34
       12.1.1.
       11.1.1. siprec Option Tag  . . . . . . . . . . . . . . . . . . 34
       12.1.2. 35
       11.1.2. record-aware Option Tag  . . . . . . . . . . . . . . . 34
     12.2. 35
     11.2. Registration of media feature tags . . . . . . . . . . . . 34
       12.2.1. 35
       11.2.1. src feature tag  . . . . . . . . . . . . . . . . . . . 34
       12.2.2. 35
       11.2.2. srs feature tag  . . . . . . . . . . . . . . . . . . . 35
     12.3. 36
     11.3. New Content-Disposition Parameter Registrations  . . . . . 35
     12.4. 36
     11.4. Media Type Registration  . . . . . . . . . . . . . . . . . 35
       12.4.1. 36
       11.4.1. Registration of MIME Type application/rs-metadata  . . 35
       12.4.2. 36
       11.4.2. Registration of MIME Type
               application/rs-metadata-request  . . . . . . . . . . . 36
     12.5. 37
     11.5. SDP Attributes . . . . . . . . . . . . . . . . . . . . . . 36
       12.5.1. 37
       11.5.1. 'record' SDP Attribute . . . . . . . . . . . . . . . . 36
       12.5.2. 37
       11.5.2. 'recordpref' SDP Attribute . . . . . . . . . . . . . . 36
   13. 37
   12. Security Considerations  . . . . . . . . . . . . . . . . . . . 37
     13.1. 38
     12.1. Authentication and Authorization . . . . . . . . . . . . . 38
     12.2. RTP handling . . . . . . . . . . . . . . . . . . . . . . . 37
     13.2. Authentication 39
     12.3. Metadata . . . . . . . . . . . . . . . . . . . . . . . . . 39
     12.4. Storage and Authorization playback . . . . . . . . . . . . . 37
   14. . . . . . . 40
   13. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 37
   15. 40
   14. References . . . . . . . . . . . . . . . . . . . . . . . . . . 38
     15.1. 40
     14.1. Normative References . . . . . . . . . . . . . . . . . . . 38
     15.2. 40
     14.2. Informative References . . . . . . . . . . . . . . . . . . 38 41
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 39 42

1.  Introduction

   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:

   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 5 provides the procedures for establishing a recording
   session between a SRC and a SRS.  Section 6 describes the SDP in a
   recording session.  Section 7 describes the RTP handling in a
   recording session.  Section 8 describes the mechanism to deliver
   recording metadata from the SRC to the SRS.

   Section 10 describes the procedures for user agents participating in
   a CS to receive recording indications and to provide preferences for
   recording.

   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
   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
   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 |
            |<----------------------------------------------------|
            |(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.  User agents that are recording-aware
   supports receiving recording indications with new SDP attribute
   a=record and the recording-aware UA can also set recording preference
   in the CS with a new 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
   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.

6.1.  Procedures at the SRC

   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.

6.2.  Procedures at

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

   When an SRS

   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.

7.

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 Handling

   The SRC extensions for receiving recording indication and SRS follows for
   requesting recording preferences for the SDP offer/answer model in [RFC3264].  The
   rest call.  A recording-aware UA
   MUST indicate that it can accept reporting of this section describes conventions used in a recording
   session.

7.1.  Procedures at indication
   provided by the SRC

   Since with a new option tag "record-aware" when
   initiating or establishing a CS, meaning including the SRC does not expect to receive media from "record-aware"
   tag in the SRS, Supported header in the SRC
   typically sets each media stream of 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

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 label name a=label attribute only applies to the same SIP message as SDP and Metadata
   conveyed in the
   SDP, meaning bodies of the SIP request or response that the label name can be reused by another media
   stream within the same recording session.
   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 of SDP offer from SRC with both audio and
   video recorded streams.  Note that the following example contain
   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
       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.

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 reason 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 streams with a=sendonly attribute.  Note that when a CS
   stream is muted/unmuted, this information is conveyed in the metadata
   by the SRC.  The SRC SHOULD NOT modify the media stream with
   a=inactive for mute since this operation is reserved for pausing the
   RS media.

7.2.  Procedures at

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 SRS

   The SRS only receives RTP streams
   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 by 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

   The SRS only receives RTP streams from the SRC, the SDP answer
   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 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 4 8
       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 4 8
       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].

   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

8.  RTP Handling

   This section provides recommendations and guidelines

7.3.  Procedures 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 Recording-aware User Agents (UAs) SHOULD utilize the
   mechanisms provided by RTP in

7.3.1.  Recording indication

   When a well-defined and predicable manner.
   It is recording-aware UA receives an SDP offer or answer that
   includes the goal of this document to make a=record attribute, the reader aware of these
   mechanisms and UA MUST 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) recording
   indication to
   allow monitoring of the data delivery.  RTCP, as defined in
   [RFC3550], end user whether the recording is on, off, or
   paused for each medium 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 most recently received a=record
   SDP attribute for RTCP that medium.

   If a call is REQUIRED, per
   [RFC3550], traversed through one or more SIP B2BUA, and it provides, among other things, the following
   important functionality happens
   that there are more than one SRC in relation the call path, the recording
   indication attribute does not provide any hint as to SIPREC:

   1) Feedback on which SRC is
   performing the quality of recording, meaning the data distribution

   This feedback from endpoint only knows that the receivers may be
   call is being recorded.  This attribute is also not used as an
   indication to diagnose faults negotiate which SRC in the distribution.  As such, RTCP is a well-defined and efficient
   mechanism for the SRS to inform the SRC, call path will perform
   recording and for the SRC is not used as a request to inform start/stop recording if
   there are multiple SRCs in the call path.

7.3.2.  Recording aware UAs, of issues that arise with respect to preference

   A participant in a CS MAY set the
   reception of media that is recording preference in the CS to
   be recorded.

   2) Carries a persistent transport-level identifier for an RTP source
   called the canonical name recorded or CNAME

   The SSRC identifier may change if a conflict is discovered not recorded at session establishment or a
   program during the
   session.  A new 'recordpref' SDP attribute is restarted; in which case receivers can use introduced, and the CNAME to
   keep track of each participant.  Receivers
   participant in CS may also use set this recording preference atrribute in any
   SDP offer/answer at session establishment time or during the CNAME session.
   The SRC is not required to
   associate multiple data streams honor the recording preference from a given
   participant in a set of
   related RTP sessions, for example to synchronize audio and video.
   Synchronization of media streams is also facilitated by based on local policies at the NTP SRC, and
   RTP timestamps included the participant
   can learn the recording indication through the a=record SDP attribute
   as described in RTCP packets by data senders.

8.1.2.  RTP Profile the above section.

   The RECOMMENDED RTP profiles for SDP a=recordpref attribute can appear at 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, media level or
   session level 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.

8.1.4.  CSRC

   The contributing source (CSRC), as defined can appear 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 SDP offer or answer.  When the SSRC identifiers of
   attribute is applied at the sources that contributed session level, the recording preference
   applies to all media stream in the
   generation of a particular packet into SDP.  When the RTP header of that packet.
   This list attribute is called
   applied at the CSRC list.  It is RECOMMENDED that a SRC or
   Recording aware UA, when acting a mixer, sets media level, the CSRC list
   accordingly, recording preference applies to the
   media stream only, and that overrides the SRC and SRS interpret recording preference if
   also set at the CSRC list
   appropriately when received.

8.1.5.  SDES session level.  The Source Description (SDES), as defined in [RFC3550], contains an
   SSRC/CSRC identifier followed user agent can change the
   recording preference by a list of zero or more items, which
   carry information about changing the SSRC/CSRC.  End systems send one SDES
   packet containing their own source identifier (the same as a=recordpref attribute in
   subsequent SDP offer or answer.  The absence of the SSRC a=recordpref
   attribute 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 SDP indicates that the UA has no recording
   preference.

   The Canonical End-Point Identifier (CNAME), as following is the ABNF of the recordpref attribute:

      attribute /= recordpref-attr

      ; attribute defined in [RFC3550],
   provides RFC 4566

      recordpref-attr = "a=recordpref:" pref

      pref = "on" / "off" / "pause" / "nopreference"

   on Sets the binding from preference to record if it has not already been started.
      If the SSRC identifier recording is currently paused, the preference is to an identifier resume
      recording.

   off  Sets the preference for no recording.  If recording has already
      been started, then the source (sender or receiver) that remains constant.  It preference is
   important to stop the SRC and Recording aware UAs generate CNAMEs
   appropriately and recording.

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

   nopreference  To indicate that the SRC and SRS interpret UA has no preference on recording.

8.  RTP Handling

   This section provides recommendations and use them for
   this purpose.  Guidelines for generating CNAME values are provided in
   "Guidelines 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 and RTCP
   in an
   inactive state for extended periods of times for any of a number the context of
   valid reasons. SIPREC.  In order for the bindings and any pinholes in NATs/
   firewalls to remain active during such intervals, it is RECOMMENDED
   that communicate most effectively,
   the SRC, SRS, Session Recording Client (SRC), the Session Recording Server
   (SRS), and any Recording aware UAs follow the keep-alive
   procedure recommended in "Application Mechanism for Keeping Alive User Agents (UAs) SHOULD utilize the
   NAT Mappings Associated to RTP/RTP Control Protocol (RTCP) Flows"
   [RFC6263] for all RTP media streams.

8.1.7.  RTCP Feedback Messages

   "Codec Control Messages
   mechanisms provided by RTP in a well-defined and predicable manner.
   It is the RTP Audio-Visual Profile with Feedback
   (AVPF)" [RFC5104] specifies extensions goal of this document to make the messages defined in
   AVPF [RFC4585].  Support for and proper usage reader aware of these messages is
   mechanisms and provide recommendations and guidelines.

8.1.  RTP Mechanisms

   This section briefly describes important to SRC, SRS, RTP/RTCP constructs and Recording aware UA implementations.  Note
   mechanisms that these messages are applicable only when using particularly useful within the AVFP or SAVPF content of SIPREC.

8.1.1.  RTCP

   The RTP profiles

8.1.7.1.  Full Intra Request

   A Full Intra Request (FIR) Command, when received data transport is augmented 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 control protocol (RTCP) 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
   allow monitoring 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), data delivery.  RTCP, as defined in [RFC4585], informs
   [RFC3550], is based on the
   encoder periodic transmission of control packets
   to all participants in the RTP session, using the same distribution
   mechanism as the loss of an undefined amount of coded video data
   belonging to one or more pictures.  Using packets.  Support for RTCP is REQUIRED, per
   [RFC3550], and it provides, among other things, the FIR command following
   important functionality in relation to recover SIPREC:

   1) Feedback on the quality of the data distribution

   This feedback from errors is explicitly disallowed, and instead the PLI message
   SHOULD be used.  FIR SHOULD receivers may be used only to diagnose faults in situations where not
   sending a decoder refresh point would render the video unusable for
   the users.  Examples where sending FIR distribution.  As such, RTCP is appropriate include a
   multipoint conference when a new user joins the conference well-defined and no
   regular decoder refresh point interval is established, efficient
   mechanism for the SRS to inform the SRC, and a video
   switching MCU that changes streams.

8.1.7.3.  Temporary Maximum Media Stream Bit Rate Request

   A receiver, translator, or mixer uses for the Temporary Maximum Media
   Stream Bit Rate Request (TMMBR) SRC to request a sender inform
   Recording aware UAs, of issues that arise with respect to limit the
   maximum bit rate for a
   reception of media stream that is to be recorded.

   2) Carries a persistent transport-level identifier for an RTP source
   called the provided value.
   Appropriate 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 TMMBR facilitates rapid adaptation
   related RTP sessions, for example to changes in
   available bandwidth.

8.1.7.3.1.  Renegotiation synchronize audio and video.
   Synchronization of SDP bandwidth attribute

   If it media streams is likely that also facilitated by the new value indicated NTP and
   RTP timestamps included in RTCP packets by TMMBR will be valid data senders.

8.1.2.  RTP Profile

   The RECOMMENDED RTP profiles for the remainder of the session, the TMMBR sender is expected to
   perform a renegotiation of the session upper limit 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 the session
   signaling protocol.  Therefore
   encrypted RTP streams, and "Extended RTP Profile for SIPREC, 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 are 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 use TMMBR events, rather
   than strictly according to bandwidth.  AVPF based codec control
   messages provide efficient mechanisms for temporary an SRC, SRS, and Recording
   aware UAs to handle events such as scene changes, error recovery, and renegotiation of
   dynamic bandwidth via SDP offer/answer for adjustments.  These messages are discussed in more permanent changes.

8.1.8.  Symmetric RTP/RTCP for Sending and Receiving

   Within an SDP offer/answer exchange, RTP entities choose the RTP
   detail later in this document.

   SAVP and
   RTCP transport addresses (i.e., IP addresses SAVPF provide media encryption, integrity protection, replay
   protection, and port numbers) on
   which to receive packets.  When sending packets, the RTP entities may
   use the same a limited form of source port authentication.  They do not
   contain or require a different specific keying mechanism.

8.1.3.  SSRC

   The synchronization source port (SSRC), as those signaled
   for receiving packets.  When the transport address used to send and
   receive RTP is the same, it defined in [RFC3550] is termed "symmetric RTP" [RFC4961].
   Likewise, when carried
   in the transport address used to send RTP header and receive in various fields of RTCP packets.  It is
   the same, it is termed "symmetric RTCP" [RFC4961].

   When sending RTP, it is REQUIRED to use symmetric RTP.  When sending
   RTCP, it a
   random 32-bit number that is REQUIRED required to use symmetric RTCP.  Although be globally unique within an SRS will not
   normally send RTP, it will send RTCP as well as receive
   RTP and RTCP.
   Likewise, although an SRC will 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 normally receive RTP from likely to choose the SRS,
   it will receive RTCP as well as send RTP and RTCP.

      Note: Symmetric RTP same number.  Guidelines regarding
   SSRC value selection and symmetric RTCP conflict resolution are different from RTP/RTCP
      multiplexing [RFC5761].

8.2.  Roles

   An SRC has the task provided in
   [RFC3550].

   The SSRC may also be used to separate different sources of gathering media from
   within a single RTP session.  For this reason as well as for conflict
   resolution, it is important that the various SRC, SRS, and Recording aware
   UAs handle changes in one or
   more Communication Sessions (CSs) SSRC values and forwarding the information to
   the SRS within properly identify the context reason of a corresponding Recording Session (RS).
   There are numerous ways
   the change.  The CNAME values carried in which an SRC may do RTCP facilitate this is, including but
   not limited to, appearing as a UA within a CS, or
   identification.

8.1.4.  CSRC

   The contributing source (CSRC), as defined in [RFC3550], identifies
   the source of a B2BUA between
   UAs within a CS.

                     (Recording Session)   +---------+
                   +------------SIP------->|         |
                   |  +------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 stream of RTP packets that has contributed to the
   combined stream produced by an RTP mixer.  The following subsections define mixer inserts a set list
   of roles an SRC may choose to
   play based on its position with respect the SSRC identifiers of the sources that contributed to the
   generation of a UA within particular packet into the RTP header of that packet.

   This list is called the CSRC list.  It is RECOMMENDED that a CS, SRC or
   Recording aware UA, when acting a mixer, sets the CSRC list
   accordingly, and that the SRC and an SRS within interpret the CSRC list
   appropriately when received.

8.1.5.  SDES

   The Source Description (SDES), as defined in [RFC3550], contains an RS.
   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 CS and mixer sends one SDES packet containing a corresponding RS
   chunk for each contributing source from which it is receiving SDES
   information, or multiple complete SDES packets if there are independent
   sessions; therefore, 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 may play a different role within a CS
   than it does within and Recording aware UAs generate CNAMEs
   appropriately and that the corresponding RS.

8.2.1. SRC acting as an and SRS interpret and use them for
   this purpose.  Guidelines for generating CNAME values are provided in
   "Guidelines for Choosing RTP Translator

   The SRC may act as a translator, as defined Control Protocol (RTCP) Canonical Names
   (CNAMEs)" [RFC6222].

8.1.6.  Keepalive

   It is anticipated that media streams in [RFC3550].  A defining
   characteristic SIPREC may exist in an
   inactive state for extended periods of times for any of a translator is that it forwards RTP packets with
   their SSRC identifier intact.  There are two types number of translators,
   one that simply forwards,
   valid reasons.  In order for the bindings and another that performs transcoding
   (e.g., from one codec to another) any pinholes in addition NATs/
   firewalls to forwarding.

8.2.1.1.  Forwarding Translator

   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 remain active during such intervals, it is RECOMMENDED
   that the SRC SRC, SRS, and MUST be sent
   separately to Recording aware UAs follow the SRS.  All 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 reports MUST be passed by Feedback Messages

   "Codec Control Messages in the SRC
   between RTP Audio-Visual Profile with Feedback
   (AVPF)" [RFC5104] specifies extensions to the UAs messages defined in
   AVPF [RFC4585].  Support for and the proper usage of these messages is
   important to SRC, SRS, such that the UAs and SRS are able to
   detect any SSRC collisions.

   RTCP Sender Reports generated by a Recording aware UA sending a stream MUST be
   forwarded to implementations.  Note
   that these messages are applicable only when using the SRS.  RTCP Receiver Reports generated AVFP or SAVPF
   RTP profiles

8.1.7.1.  Full Intra Request

   A Full Intra Request (FIR) Command, when received by the SRS
   MUST be forwarded 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 relevant UA.

   UAs may receive multiple sets encoding process of RTCP Receiver Reports, one or more
   from other UAs participating any subsequent picture sent
   in the CS, and one from the SRS
   participating 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 RS.  A Recording aware UA SHOULD be prepared to
   process the RTCP Receiver Reports from available bit
   rate is small, the SRS, whereas a recording
   unaware UA may discard such RTCP packets as not use of relevance.

   If SRTP a decoder refresh point implies a delay
   that 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 significantly longer than the same keys 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 used, it need not occur.
   Section 13 provides additional information on SRTP and keying
   mechanisms.

   If packet loss occurs, either from the UA to the SRC or
   discouraged from using the SRC
   to the SRS, the SRS method described except for backward
   compatibility purposes.  Implementations 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 use FIR messages
   instead.

8.1.7.2.  Picture Loss Indicator

   Picture Loss Indication (PLI), as a transcoding translator, defined in [RFC4585], informs the
   encoder of the loss of an SRC MAY perform
   transcoding (e.g., from undefined amount of coded video data
   belonging to one codec or more pictures.  Using the FIR command to another), recover
   from errors is explicitly disallowed, and this may result instead the PLI message
   SHOULD be used.  FIR SHOULD be used only in situations where not
   sending a different rate of packets between what decoder refresh point would render the SRC receives and what video unusable for
   the SRC sends.  As users.  Examples where sending FIR is appropriate include a
   multipoint conference 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 new user joins the SRC conference and
   MUST be sent separately to the SRS.  All RTCP reports MUST be passed
   by the SRC between the UAs no
   regular decoder refresh point interval is established, and the SRS, such a video
   switching MCU that changes streams.

8.1.7.3.  Temporary Maximum Media Stream Bit Rate Request

   A receiver, translator, or mixer uses the UAs and SRS are
   able Temporary Maximum Media
   Stream Bit Rate Request (TMMBR) to detect any SSRC collisions.

   RTCP Sender Reports generated by request a UA sending sender to limit the
   maximum bit rate for a media stream MUST be
   forwarded to the SRS.  RTCP Receiver Reports generated 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 SRS
   MUST be forwarded to session, the relevant UA.  The SRC may need TMMBR sender is expected to manipulate
   perform a renegotiation of the RTCP Receiver Reports session upper limit using the session
   signaling protocol.  Therefore for SIPREC, implementations are
   RECOMMENDED to take account of any transcoding that has
   taken place.

   UAs may receive multiple sets use TMMBR for temporary changes, and renegotiation of RTCP Receiver Reports, one or
   bandwidth via SDP offer/answer for more
   from other UAs participating in the CS, permanent changes.

8.1.8.  Symmetric RTP/RTCP for Sending and one from the SRS
   participating in Receiving

   Within an SDP offer/answer exchange, RTP entities choose the RS.  A Recording aware UA SHOULD be prepared RTP and
   RTCP transport addresses (i.e., IP addresses and port numbers) on
   which to
   process receive packets.  When sending packets, the RTCP Receiver Reports from RTP entities may
   use the SRS, whereas same source port or a recording
   unaware UA may discard such RTCP packets different source port as not of relevance.

   If SRTP is those signaled
   for receiving packets.  When the transport address used on both to send and
   receive RTP is the CS same, it is termed "symmetric RTP" [RFC4961].
   Likewise, when the transport address used to send and receive RTCP is
   the RS, decryption and/or re-
   encryption may occur.  For example, if different keys are used, 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 occur.  If 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 same keys are used, SRS,
   it need not occur.
   Section 13 provides additional information on SRTP will receive RTCP as well as send RTP and keying
   mechanisms.

   If packet loss occurs, either RTCP.

      Note: Symmetric RTP and symmetric RTCP are different from RTP/RTCP
      multiplexing [RFC5761].

8.2.  Roles

   An SRC has the UA to task of gathering media from the SRC various UAs in one or from
   more Communication Sessions (CSs) and forwarding the SRC information to
   the SRS, 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------->|         |
                   |  +------RTP/RTCP----->|   SRS SHOULD detect   |
                   |  |    +-- 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 attempt to recover from the
   loss.  The a corresponding RS are independent
   sessions; therefore, an SRC does not may play a different role in this other within a CS
   than forwarding it does within the
   associated RTP and RTCP packets.

8.2.2. corresponding RS.

8.2.1.  SRC acting as an RTP Mixer

   In the case of the Translator

   The SRC acting may act as a RTP mixer, translator, 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 [RFC3550].  A defining
   characteristic of a mixer can perform transcoding, and can
   even cope translator is that it forwards RTP packets with different codings received from different UAs.  RTCP
   Sender Reports and Receiver Reports are not forwarded by an SRC
   acting as mixer, but there
   their SSRC identifier intact.  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 two types of SRTP between the UAs and SRC for the CS.  Section 13
   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 translators,
   one that simply forwards, and another that performs transcoding
   (e.g., from the
   SRC one codec to the SRS, the SRS SHOULD detect and attempt another) in addition to recover from the
   loss.

8.2.3.  SRC forwarding.

8.2.1.1.  Forwarding Translator

   When acting as an a forwarding translator, RTP Endpoint

   The case of received as separate
   streams from different sources (e.g., from different UAs with
   different SSRCs) cannot be mixed by the SRC acting as an RTP endpoint, as defined in
   [RFC3550], is similar and MUST be sent
   separately to the mixer case, except that SRS.  All RTCP reports MUST be passed by the RTP session SRC
   between the SRC UAs and the SRS is considered completely independent from
   the RTP session SRS, such that is part of the CS.  The SRC can, but need not,
   mix RTP streams from different participants prior 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 between the SRC and Receiver Reports generated by the SRS is completely independent of
   RTCP on
   MUST be forwarded to the CS.

   The use relevant UA.

   UAs may receive multiple sets of SRTP between RTCP Receiver Reports, one or more
   from other UAs participating in the SRC CS, and one from the SRS for
   participating in the RS is independent
   of RS.  A Recording aware UA SHOULD be prepared to
   process the use RTCP Receiver Reports from the SRS, whereas a recording
   unaware UA may discard such RTCP packets as not of relevance.

   If SRTP between is used on both the UAs CS and SRC for the CS. 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 13 12 provides additional information on SRTP and keying
   mechanisms.

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

8.3.  RTP Session Usage by  The SRC

   There are multiple ways that 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 choose to deliver recorded
   media MAY perform
   transcoding (e.g., from one codec to an SRS.  In some cases, it another), and this may use a single RTP session for
   all media within the RS, whereas result in others it may use multiple RTP
   sessions.  The following subsections provide examples of basic RTP
   session usage by the SRC, including
   a discussion different rate of how packets between what the RTP
   constructs and mechanisms covered previously are used.  An SRC may
   choose to use one or more of receives and what
   the RTP session usages within SRC sends.  As when acting as a single
   RS.  The set of forwarding translator, RTP session usages described is not meant to
   received as separate streams from different sources (e.g., from
   different UAs with different SSRCs) cannot be
   exhaustive.

8.3.1. mixed by the SRC Using Multiple m-lines

   When using multiple m-lines, an and
   MUST be sent separately to the SRS.  All RTCP reports MUST be passed
   by the SRC includes each m-line in an SDP
   offer 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.  The SDP answer from  RTCP Receiver Reports generated by the SRS
   MUST include all
   m-lines, with any rejected m-lines indicated with a zero port, per
   [RFC3264].  Having received the answer, be forwarded to the relevant UA.  The SRC starts sending media may need to manipulate
   the SRS as indicated 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 answer.  Alternatively, if the SRC
   deems CS, and one from the level of support indicated SRS
   participating in the answer to RS.  A Recording aware UA SHOULD be
   unacceptable, it may initiate another SDP offer/answer exchange in
   which an alternative RTP session usage is negotiated.

   In order prepared to preserve
   process the mapping RTCP Receiver Reports from the SRS, whereas a recording
   unaware UA may discard such RTCP packets as not of media to participant within relevance.

   If SRTP is used on both the
   CSs in CS and the RS, decryption and/or re-
   encryption may occur.  For example, if different keys are used, it
   will occur.  If the SRC SHOULD map each unique CNAME within 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 CSs UA to
   a unique CNAME within the RS.  Additionally, the SRC SHOULD map each
   unique combination of CNAME/SSRC within or from the CSs SRC
   to a unique CNAME/
   SSRC within the RS.  In doing to, SRS, the SRS SHOULD detect and attempt to recover from the
   loss.  The SRC may act as an does not play a role in this other than forwarding the
   associated RTP
   translator or and RTCP packets.

8.2.2.  SRC acting as an RTP endpoint.

   The following figure illustrates a Mixer

   In the case in which each UA represents of the SRC acting as a
   participant contributing two RTP sessions (e.g. one for audio and one
   for video), each with a single SSRC.  The SRC acts mixer, as an defined in
   [RFC3550], the SRC combines RTP
   translator streams from different UA and delivers the media to sends
   them towards the SRS using four RTP sessions,
   each with a single its own SSRC.  The CNAME and SSRC values used by SSRCs from the UAs
   contributing UA SHOULD be conveyed as CSRCs identifiers within their media streams are preserved in this
   stream.  The SRC may make timing adjustments among the media received
   streams from and generate its own timing on the SRC stream sent to the SRS.

                                                         +---------+
                                 +------------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 SSRC Multiplexing

   When using SSRC multiplexing,
   Optionally an SRC multiplexes RTP packets of the
   same media type from multiple RTP sessions into acting as a single RTP session mixer can perform transcoding, and can
   even cope with multiple SSRC values.  The SRC includes one m-line for each RTP
   session in different codings received from different UAs.  RTCP
   Sender Reports and Receiver Reports are not forwarded by an SDP offer to 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 SDP answer from use of SRTP between the SRC and the SRS MUST
   include all m-lines, with any rejected m-lines indicated with for the
   zero port, per [RFC3264].  Having received RS is independent
   of the answer, use of SRTP between the UAs and SRC starts
   sending media for the CS.  Section 12
   provides additional information on SRTP and keying mechanisms.

   If packet loss occurs from the UA to the SRS as indicated in SRC, the answer.

   In order SRC SHOULD detect
   and attempt to preserve recover from the mapping of media loss.  If packet loss occurs from the
   SRC to participant within the
   CSs in SRS, the RS, SRS SHOULD detect and attempt to recover from the
   loss.

8.2.3.  SRC SHOULD map each unique combination acting as an RTP Endpoint

   The case of CNAME/
   SSRC within the CSs SRC acting as an RTP endpoint, as defined in
   [RFC3550], is similar to a unique SSRC within the RS.  The CNAMEs used
   in mixer case, except that the CSs are not preserved within RTP session
   between the SRC and the RS.  The SRS relies on is considered completely independent from
   the
   SIPREC metadata to determine RTP session that is part of the participants included within each
   multiplexed stream. CS.  The SRC MUST avoid SSRC collisions, rewriting
   SSRCs if necessary.  In doing to, can, but need not,
   mix RTP streams from different participants prior to sending to the
   SRS.  RTCP between the SRC acts as an RTP endpoint.

   In and the SRS is completely independent of
   RTCP on the CS.

   The use of SRTP between the event SRC and the SRS does not support SSRC multiplexing, for the SRC
   becomes aware RS is independent
   of this when it receives RTCP receiver reports from the
   SRS indicating the absence use of any packets SRTP between the UAs and SRC for one or more of the
   multiplexed SSRC values. CS.  Section 12
   provides additional information on SRTP and keying mechanisms.

   If packet loss occurs from the SRC deems UA to the level of support
   indicated in SRC, the RTCP receiver report to be unacceptable, it may
   initiate another SDP offer/answer exchange in which an alternative
   RTP session usage is negotiated.

   The following figure illustrates a case in which each UA represents a
   participant contributing two RTP sessions (e.g. one for audio and
   another for video), each with a single SSRC.  The SRC delivers SHOULD detect
   and attempt to recover from the
   media loss.  If packet loss occurs from the
   SRC to the SRS, the SRS using two RTP sessions, multiplexing one stream with SHOULD detect and attempt to recover from the same
   loss.

8.3.  RTP Session Usage by SRC

   There are multiple ways that an SRC may choose to deliver recorded
   media type from each participant into to an SRS.  In some cases, it may use a single RTP session
   containing two SSRCs.  The SRC uses its own CNAME and SSRC values,
   but it preserves for
   all media within the mapping RS, whereas in others it may use multiple RTP
   sessions.  The following subsections provide examples of unique CNAME/SSRC used basic RTP
   session usage by the UAs
   within their media streams in the media streams from 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
   SRS.

                                                         +---------+
                                                         |         |
                                   +-----SSRC SAa,SBa--->|         |
                                   |   +-SSRC SAv,SBv--->|   SRS   |
                                   |   |                 |         |
                                   |   |                 +---------+
                                   |   |
                                   |   |
        +---------+             +----------+             +---------+
        |         |---SSRC Aa-->|   SRC    |<--SSRC Ba---|         |
        |  UA-A   |             |(CNAME-S) |             |  UA-B   |
        |(CNAME-A)|---SSRC Av-->|          |<--SSRC Bv---|(CNAME-B)|
        +---------+             +----------+             +---------+

                  Figure 10: SRC Using SSRC Multiplexing

8.3.3. RTP session usages within a single
   RS.  The set of RTP session usages described is not meant to be
   exhaustive.

8.3.1.  SRC Using Mixing Multiple m-lines

   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 multiple m-lines, an SRC includes one m-line for each RTP session 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 the 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.  The SRC MUST avoid SSRC collisions, rewriting
   SSRCs if necessary when used as CSRCs in the RS.  In doing to, the SRC acts may act 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.
   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 mixer
   translator and delivers the media to the SRS using two four RTP sessions, mixing
   media from
   each participant into a single RTP session containing with a single SSRC SSRC.  The CNAME and two CSRCs. SSRC Sa values used by the UAs
   within their media streams are preserved in the media streams from
   the SRC to the SRS.

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

                   Figure 11: 9: SRC Using Mixing

9.  Metadata

9.1.  Procedures at the Multiple m-lines

8.3.2.  SRC

   The Using SSRC Multiplexing

   When using SSRC multiplexing, an SRC MUST deliver metadata to multiplexes RTP packets of the SRS in
   same media type from multiple RTP sessions into a recording session; the
   timing of which single RTP session
   with multiple SSRC values.  The SRC sends the metadata depends on when includes one m-line for each RTP
   session in an SDP offer to the metadata
   becomes available.  Metadata SHOULD be provided by SRS.  The SDP answer from the SRC in SRS MUST
   include all m-lines, with any rejected m-lines indicated with the
   initial INVITE request when establishing
   zero port, per [RFC3264].  Having received the recording session, and
   subsequent metadata updates can be provided by answer, the SRC in reINVITE
   and UPDATE requests ([RFC3311]) and responses in starts
   sending media to the recording
   session.  There are cases that metadata is not available SRS as indicated in the
   initial INVITE request sent by the SRC, for example, when a recording
   session is established in answer.

   In order to preserve the absence mapping of a communication session, and
   the SRC would update media to participant within the recording session with metadata whenever
   metadata becomes available.

   Certain metadata attributes are contained
   CSs in the SDP, and others are
   contained in a new content type "application/rs-metadata".  The
   format RS, the SRC SHOULD map each unique combination of CNAME/
   SSRC within the metadata is described as part of CSs to a unique SSRC within the mechanism RS.  The CNAMEs used
   in
   [I-D.ietf-siprec-metadata].  A new "disposition-type" of Content-
   Disposition is defined for the purpose of carrying metadata and CSs are not preserved within the
   value is "recording-session". RS.  The "recording-session" value
   indicates that SRS relies on the "application/rs-metadata" content contains
   SIPREC metadata to be handled by the SRS, and the disposition can be carried
   in either INVITE or UPDATE requests or responses sent by determine the SRC.

   Metadata sent by participants included within each
   multiplexed stream.  The SRC MUST avoid SSRC collisions, rewriting
   SSRCs if necessary.  In doing to, the SRC can be categorized acts as either a full metadata
   snapshot or partial update.  A full metadata snapshot describes all an RTP endpoint.

   In the recorded streams and all metadata associated with event the recording
   session.  When SRS does not support SSRC multiplexing, the SRC sends a full metadata snapshot,
   becomes aware of this when it receives RTCP receiver reports from the SRC MUST
   send an INVITE
   SRS indicating the absence of any packets for one or an UPDATE request ([RFC3311]) with an SDP offer and more of the "recording-session" disposition.  A partial update represents an
   incremental update since
   multiplexed SSRC values.  If the last metadata update sent by SRC deems the level of support
   indicated in the RTCP receiver report to be unacceptable, it may
   initiate another SDP offer/answer exchange in which an alternative
   RTP session usage is negotiated.

   The following figure illustrates a case in which each UA represents a
   participant contributing two RTP sessions (e.g. one for audio and
   another for video), each with a single SSRC.  The SRC delivers the SRC.  A
   partial update sent by
   media to the SRC can be an INVITE request or response SRS using two RTP sessions, multiplexing one stream with an SDP offer, or an INVITE/UPDATE request or response containing
   the same media type from each participant into a "recording-session" disposition, or an INVITE request single RTP session
   containing
   both an SDP offer two SSRCs.  The SRC uses its own CNAME and SSRC values,
   but it preserves the "recording-session" disposition.

   The following is an example mapping of a full metadata snapshot sent unique CNAME/SSRC used by the
   SRC UAs
   within their media streams 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-098392474
      To: <sip:recorder@example.com>
      Call-ID: d253c800-b0d1ea39-4a7dd-3f0e20a
      CSeq: 101 INVITE
      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] media streams from the SRC to the
   SRS.

                                                         +---------+
                                                         |         |
                                   +-----SSRC SAa,SBa--->|         |
                                   |   +-SSRC SAv,SBv--->|   SRS   |
                                   |   |                 |         |
                                   |   |                 +---------+
                                   |   |
                                   |   |
        +---------+             +----------+             +---------+
        |         |---SSRC Aa-->|   SRC    |<--SSRC Ba---|         |
        |  UA-A   |             |(CNAME-S) |             |  UA-B   |
        |(CNAME-A)|---SSRC Av-->|          |<--SSRC Bv---|(CNAME-B)|
        +---------+             +----------+             +---------+

                  Figure 12: Sample INVITE request for 10: SRC Using SSRC Multiplexing

8.3.3.  SRC Using Mixing

   When using mixing, the recording session

9.2.  Procedures at SRC combines RTP streams from different
   participants and sends them towards the SRS using its own SSRC.  The SRS receives metadata updates
   SSRCs from the contributing participants SHOULD be conveyed as CSRCs
   identifiers.  The SRC includes one m-line for each RTP session in INVITE and UPDATE
   requests.  Since the SRC can send partial updates based on the
   previous update, the SRS needs an
   SDP offer to keep track of the sequence of
   updates SRS.  The SDP answer from the SRC.

   In SRS MUST include all
   m-lines, with any rejected m-lines indicated with the case of an internal failure at zero port, per
   [RFC3264].  Having received the SRS, answer, the SRS may fail SRC starts sending media
   to
   recognize a partial update from the SRC.  The SRS may be able as indicated in the answer.

   In order to
   recover from preserve the internal failure by requesting for a full metadata
   snapshot from mapping of media to participant within the SRC.  Certain errors, such as syntax errors or
   semantic errors
   CSs in the metadata information, are likely caused by an
   error on RS, the SRC side, and it is likely SHOULD map each unique CNAME within the same error will occur
   again even when a full metadata snapshot is requested.  In order CSs to
   avoid repeating
   a unique CNAME within the same error, RS.  Additionally, the SRS can simply terminate SRC SHOULD map each
   unique combination of CNAME/SSRC within the
   recording session when CSs to a syntax error or semantic error is detected
   in unique CNAME/
   SSRC within the metadata.

   When RS.  The SRC MUST avoid SSRC collisions, rewriting
   SSRCs if necessary when used as CSRCs in the SRS explicitly requests for a full metadata snapshot, RS.  In doing to, the
   SRS MUST send an UPDATE request without an SDP offer.  A metadata
   snapshot request contains a content with
   SRC acts as an RTP mixer.

   In the content disposition type
   "recording-session".  Note that event the SRS MAY generate an INVITE
   request without an SDP offer but does not support this MUST NOT include a metadata
   snapshot request.  The format usage of CSRC values, it
   relies entirely on the content is "application/
   rs-metadata-request", and the body format is chosen SIPREC metadata to be a simple
   text-based format. determine the participants
   included within each mixed stream.

   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 13: Metadata Request 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 SRS MAY include SRC acts as an RTP mixer
   and delivers the reason why a metadata snapshot request is
   being made 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--->|         |
                                   |                     |         |
                                   |       SSRC Sa       |   SRS   |
                                   |   +---CSRC Av,Bv--->|         |
                                   |   |                 +---------+
                                   |   |
                                +----------+
        +---------+             |   SRC in the reason line.  This reason line is free
   form text, mainly designed for logging purposes on    |             +---------+
        |         |---SSRC Aa-->|(CNAME-S, |<--SSRC Ba---|         |
        |  UA-A   |             | CNAME-A, |             |  UA-B   |
        |(CNAME-A)|---SSRC Aa-->| CNAME-B) |<--SSRC Bv---|(CNAME-B)|
        +---------+             +----------+             +---------+

                        Figure 11: SRC Using Mixing

9.  Metadata

9.1.  Procedures at the SRC side.

   The
   processing of the content by the SRC is entirely optional since the
   content is for logging only, and MUST deliver metadata to the snapshot request itself is
   indicated by SRS in a recording session; the use
   timing of the application/rs-metadata-request content
   type.

   When the which SRC receives sends the request for a metadata snapshot, depends on when the SRC
   MUST provide a full metadata snapshot
   becomes available.  Metadata SHOULD be provided by the SRC in a separate the
   initial INVITE or UPDATE
   transaction, along with an SDP offer.  All request when establishing the recording session, and
   subsequent metadata updates sent can be provided by the SRC MUST be based on the new metadata snapshot.

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 reINVITE
   and UPDATE requests ([RFC3311]) and responses in the recording
   session.  There are cases that metadata is a specific use case outlined in REQ-005 or
   Use Case 4 not available in [RFC6341], where the
   initial INVITE request sent by the SRC, for example, when a recording
   session can be is 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 session, 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 would update the timing of recording session with metadata whenever
   metadata becomes available.

   Certain metadata attributes are contained in the CS. SDP, and others are
   contained in a new content type "application/rs-metadata".  The SRC only
   needs to update
   format of the metadata when new communication sessions are
   created.

   When there is no communication sessions running on described as part of the devices with
   persistent recording, there mechanism in
   [I-D.ietf-siprec-metadata].  A new "disposition-type" of Content-
   Disposition is no recorded media to stream from the
   SRC to defined for the SRS.  In certain environments where Network Address
   Translator (NAT) is used, typically a minimum purpose of flow activity is
   required to maintain carrying metadata and the NAT binding for each port opened.  Agents
   value is "recording-session".  The "recording-session" value
   indicates that support Interactive Connectivity Establishment (ICE) solves this
   problem.  For non-ICE agents, in order not the "application/rs-metadata" content contains
   metadata to lose be handled by the NAT bindings
   for SRS, and the RTP/RTCP ports opened for disposition can be carried
   in either INVITE or UPDATE requests or responses sent by the recorded streams, SRC.

   Metadata sent by the SRC and
   SRS SHOULD follow can be categorized as either a full metadata
   snapshot or partial update.  A full metadata snapshot describes all
   the recommendations provided in [RFC6263] to
   maintain recorded streams and all metadata associated with the NAT bindings.

11.  Extensions for Recording-aware User Agents

   The following sections describe recording
   session.  When the SIP and SDP extensions for
   recording-aware user agents.  A recording-aware user agent is SRC sends a
   participant in the CS that supports full metadata snapshot, the SIP and SRC MUST
   send an INVITE or an UPDATE request ([RFC3311]) with an SDP extensions for
   receiving recording indication offer and for requesting recording
   preferences for
   the call.

11.1.  Procedures at "recording-session" disposition.  A partial update represents an
   incremental update since the last metadata update sent by the record-aware user agent SRC.  A recording-aware UA MUST indicate that it can accept reporting of
   recording indication provided
   partial update sent by the SRC can be an INVITE request or response
   with a new option tag
   "record-aware" when initiating an SDP offer, or establishing an INVITE/UPDATE request or response containing
   a CS, meaning
   including "recording-session" disposition, or an INVITE request containing
   both an SDP offer and the "record-aware" tag in "recording-session" disposition.

   The following is an example of a full metadata snapshot sent by the Supported header
   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-098392474
      To: <sip:recorder@example.com>
      Call-ID: d253c800-b0d1ea39-4a7dd-3f0e20a
      CSeq: 101 INVITE
      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 12: Sample INVITE request or response.  A recording-aware UA that has
   indicated for the recording awareness MUST provide session

9.2.  Procedures at recording indication to the end user through an appropriate user interface an indication
   whether recording is on or off for a given medium SRS

   The SRS receives metadata updates from the SRC in INVITE and UPDATE
   requests.  Since the SRC can send partial updates based on the most
   recently received a=record SDP attribute for that medium.

   Some user agents that are automatons (e.g.  IVR, media server, PSTN
   gateway)
   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 not have fail to
   recognize a user interface partial update from the SRC.  The SRS may be able to render recording
   indication.  When such user agent indicates recording awareness,
   recover from the
   UA SHOULD render recording indication through other means, internal failure by requesting for a full metadata
   snapshot from the SRC.  Certain errors, such as
   passing syntax errors or
   semantic errors in the metadata information, are likely caused by an inband tone
   error on the PSTN gateway, putting 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
   indication in session when a log file, syntax error or raising an application event semantic error is detected
   in the metadata.

   When the SRS explicitly requests for a
   VoiceXML dialog.  These user agents 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 also choose not to indicate
   recording awareness, thereby relying on whatever mechanism generate an SRC
   chooses to indicate recording, such as playing INVITE
   request without an SDP offer but this MUST NOT include a tone inband.

11.1.1.  Recording preference

   A participant in metadata
   snapshot request.  The format of the content is "application/
   rs-metadata-request", and the body format is chosen to be a CS 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 13: Metadata Request

   The SRS MAY set include the recording preference reason why a metadata snapshot request is
   being made to the SRC in the CS to
   be recorded or not recorded at session establishment or during reason line.  This reason line is free
   form text, mainly designed for logging purposes on the
   session. SRC side.  The recording-aware UA sets
   processing of the indication 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 recording
   preference in the application/rs-metadata-request content
   type.

   When the SRC receives the request for a new SDP attribute a=recordpref in metadata snapshot, the CS SRC
   MUST provide a full metadata snapshot in any a separate INVITE or UPDATE
   transaction, along with an SDP
   offer/answer.  This indication of recording preference can be offer.  All subsequent metadata
   updates sent at
   session establishment time or during by the session.  The SRC is not
   required to honor the recording preference from a participant MUST be based on local policies at the SRC; the participant gets new metadata snapshot.

9.2.1.  Formal Syntax

   The formal syntax for the recording
   indication through application/rs-metadata-request MIME is
   described below using the a=record SDP attribute augmented Backus-Naur Form (BNF) as
   described in the next
   section.

   The SDP a=recordpref attribute [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 appear at be established
   in the absence of a communication session.  The SRC continuously
   records media level or in a recording session level and can appear to the SRS even in an 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 or answer.  When the
   attribute when a new communication session is applied at
   created and also does not impact the session level, timing of the recording preference
   applies CS.  The SRC only
   needs to all media stream in update the SDP. metadata when new communication sessions are
   created.

   When the attribute there is
   applied at no communication sessions running on the devices with
   persistent recording, there is no recorded 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 from the a=recordpref attribute in
   subsequent SDP offer or answer.  If
   SRC to the a=recordpref attribute SRS.  In certain environments where Network Address
   Translator (NAT) is
   omitted, then the recording preference would be assumed used, typically a minimum of flow activity is
   required to be maintain the
   recording preference set NAT binding for each port opened.  Agents
   that support Interactive Connectivity Establishment (ICE) solves this
   problem.  For non-ICE agents, in a previous SDP offer or answer.

   The following is order not to lose the ABNF of NAT bindings
   for the recordpref attribute:

      attribute /= recordpref-attr

      ; attribute defined in RFC 4566

      recordpref-attr = "a=recordpref:" pref

      pref = "on" / "off" / "pause" / "nopreference"

   on Sets RTP/RTCP ports opened for the preference to record if it has not already been started.
      If recorded streams, the recording is currently paused, SRC and
   SRS SHOULD follow the preference is recommendations provided in [RFC6263] to resume
      recording.

   off  Sets
   maintain the preference NAT bindings.

11.  IANA Considerations

11.1.  Registration of Option Tags

   This specification registers two option tags.  The required
   information for no recording.  If recording has already
      been started, then the preference this registration, as specified in [RFC3261], is to stop as
   follows.

11.1.1.  siprec Option Tag

      Name: siprec

      Description: This option tag is for identifying the recording.

   pause  If SIP session
      for the purpose of recording session only.  This is currently typically not
      used in progress, sets a Supported header.  When present in a Require header in a
      request, it indicates that the preference
      to pause UAS MUST be either a SRC or SRS
      capable of handling the recording.

   nopreference  To contexts of a recording session.

11.1.2.  record-aware Option Tag

      Name: record-aware

      Description: This option tag is to indicate that the UA has no preference on recording.

11.2.  Procedures at ability for the SRC

   The SRC MUST provide recording indication
      user agent to all participants receive recording indicators in the
   CS. media level or
      session level SDP.  When present in a UA has indicated that Supported header, it is recording-aware through the
   "record-aware" option tag,
      indicates that the SRC MUST provide UA can receive recording indications indicators in the media
      level or session level SDP.

11.2.  Registration of media feature tags

   This document registers two new SDP a=record attribute described media feature tags in the following section.
   In SIP tree
   per the absence 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 "record-aware" option tag, meaning media feature indicated by this tag: This feature
      tag indicates that the UA user agent is not recording-aware, an SRC MUST provide recording indications
   through other means such as playing a tone inband, if Session Recording Client
      for the SRC is
   required to do so (e.g. based on policies).

11.2.1. purpose for Recording indication

   While there are existing mechanisms Session.

      Values appropriate for providing an indication that
   a CS use with this feature tag: boolean

      The feature tag is being recorded, these mechanisms are usually delivered on intended primarily for use in the
   CS media streams such as playing an in-band tone following
      applications, protocols, services, or an announcement
   to the participants.  A new SDP attribute negotiation mechanisms: This
      feature tag is introduced to allow only useful for a
   recording-aware UA to render recording indication at the user
   interface.

   The 'record' SDP attribute appears at Recording Session.

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

      Security Considerations: Security considerations for this media level or session
   level
      feature tag are discussed in either SDP offer or answer.  When Section 11.1 of RFC 3840.

11.2.2.  srs feature tag

      Media feature tag name: sip.srs

      ASN.1 Identifier: 26

      Summary of the attribute is applied
   at media feature indicated by this tag: This feature
      tag indicates that the session level, user agent is a Session Recording Server
      for the indication applies to all media streams purpose for Recording Session.

      Values appropriate for use with this feature tag: boolean

      The feature tag is intended primarily for use in the SDP.  When the attribute following
      applications, protocols, services, or negotiation mechanisms: This
      feature tag is applied at the media level, only useful for a Recording Session.

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

      Security Considerations: Security considerations for this media stream only, and that overrides
      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
   indication if also set at body describes the session level.  Whenever metadata information about
   the recording
   indication needs to change, such as termination session

11.4.  Media Type Registration

11.4.1.  Registration of recording, then MIME Type application/rs-metadata

   This document registers the SRC MUST initiate a reINVITE or UPDATE application/rs-metadata MIME media type
   in order to update describe the SDP a=record
   attribute.

   The following recording session metadata.  This media type
   is defined by the ABNF 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 'record' attribute:

      attribute /= record-attr

      ; attribute defined in RFC 4566

      record-attr = "record:" indication

      indication = "on" / "off" / "paused"

   on Recording is application/rs-metadata-request MIME
   media type in progress.

   off  No order to describe a recording is in progress.

   paused  Recording is in progress by session metadata snapshot
   request.  This media type is paused.

   If a call is traversed through one defined by the following information:

   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 more SIP B2BUA, and it happens
   that there are more than one SRC in the call path, media level

   Subject to charset: no

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

   Allowed attribute does not provide any hint as 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 which SRC is
   performing the recording, meaning the endpoint only knows that the
   call is being recorded. charset: no
   This attribute is also not used as an
   indication to negotiate which SRC in provides 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.

11.2.2.  Recording preference

   When the SRC receives for the a=recordpref SDP in an SDP offer session or answer,
   the SRC chooses to honor
   media stream.

   Allowed attribute values: on, off, pause, nopreference

12.  Security Considerations

   The recording session is fundamentally a standard SIP dialog
   [RFC3261], therefore, the preference to record based on local
   policy at recording session can reuse any of the SRC.  When
   existing SIP security mechanism available for securing, session
   signaling, the SRC honors recorded media as well as metadata.  The use cases and
   requirements document [RFC6341] outlines the preference, general security
   considerations, and the following describe specific security
   recommendations.

   The SRC and SRS MUST
   also update the a=record attribute to indicate the current state of
   the recording (on/off/paused).

12.  IANA Considerations
12.1.  Registration of Option Tags

   This specification registers two option tags. support SIPS and TLS as per [RFC5630].  The required
   information for this registration,
   Recording Session SHOULD be at least as specified in [RFC3261], is secure as
   follows.

12.1.1.  siprec Option Tag

      Name: siprec

      Description: This option tag is for identifying the SIP session
      for Communication
   Session, meaning using at least the purpose same strength of recording session only.  This cipher suite as
   the CS if the CS is typically secured.  For example, if the CS uses SIPS for
   signalling and RTP/SAVP for media, then the RS does not
      used in a Supported header.  When present in a Require header downgrade the
   level of security in a
      request, it indicates that the UAS MUST be either a SRC 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
      capable of handling are in the contexts same administrative domain and
   the same physical switch that prevents outside user access, some SRC
   may choose lower the level of a security when establishing the
   recording session.

12.1.2.  record-aware Option Tag

      Name: record-aware

      Description: This option tag is to indicate  While physically securing the ability for SRC and SRS may
   prevent an outside attacker from accessing important call recordings,
   this still does not prevent from an inside attacker from accessing
   the
      user agent internal network to gain access to receive recording indicators in media level or
      session level SDP.  When present in a Supported header, it
      indicates that the UA can receive call recordings.

12.1.  Authentication and Authorization

   The recording indicators in media
      level or session level SDP.

12.2.  Registration of media feature tags

   This document registers two new media feature tags in reuses the SIP mechanism to challenge requests
   that is based on HTTP authentication.  The mechanism relies on 401
   and 407 SIP responses as well as other SIP header fields for carrying
   challenges and credentials.

   At the SIP tree
   per transport level, the process defined in [RFC2506] and [RFC3840]

12.2.1.  src feature tag

      Media feature tag name: sip.src

      ASN.1 Identifier: 25

      Summary of recording session uses TLS authentication
   to validate the media feature indicated by this tag: This feature
      tag indicates that authenticity of the user agent is a Session Recording Client SRC and SRS.  The SRC and SRS
   MUST implement TLS mutual authentication for establishing the purpose for Recording Session.

      Values appropriate for
   recording session, and whether the SRC/SRS chooses to use with this feature tag: boolean

      The feature tag
   authentication is intended primarily for use a deployment decision.  In deployments where the
   SRC and the SRS are in the following
      applications, protocols, services, same administrative domain, the deployment
   may choose not to authenticate each other 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 have SRC
   authenticate the SRS as there is an inherent trust relation between
   the SRC and the SRS when they are discussed hosted in Section 11.1 of RFC 3840.

12.2.2.  srs feature tag

      Media feature tag name: sip.srs

      ASN.1 Identifier: 26

      Summary of the media feature indicated by this tag: This feature
      tag indicates that same administrative
   domain.  In deployments where the user agent is SRS can be hosted on a Session Recording Server
      for the purpose for Recording Session.

      Values appropriate for use with this feature tag: boolean

      The feature tag different
   administrative domain, then it is intended primarily for use in important to perform mutual
   authentication to ensure the following
      applications, protocols, services, or negotiation mechanisms: This
      feature tag is only useful for a Recording Session.

      Examples authenticity of both the SRC and the SRS
   before transmitting any recorded media.  The risk of typical use: Routing not
   authenticating the request SRS is that the recording may be sent to a Session
      Recording Client.

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

12.3.  New Content-Disposition Parameter Registrations

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

   recording-session
   compromised SRS and that sensitive call recording will be obtained by
   an attacker.  On the body describes other hand, the metadata information about risk of not authenticating the
   SRC is that an SRS will be willingly accept any call recording session

12.4.  Media Type Registration

12.4.1.  Registration from
   an unknown SRC and allow potential forgery of MIME Type application/rs-metadata

   This document registers the application/rs-metadata MIME media type
   in order call recordings.

   The SRS may have its own set of recording policies to describe authorize
   recording requests from the SRC.  The use of recording session metadata.  This media type policies is defined by
   outside the following information:

   Media type name: application

   Media subtype name: rs-metadata
   Required parameters: none

   Options parameters: none

12.4.2.  Registration scope 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 Session Recording Protocol.

12.2.  RTP handling

   In many scenarios it will be critical that the media type is defined by transported
   between the following information:

   Media type name: application SRC and SRS to be protected.  Media subtype name: rs-metadata-request

   Required parameters: none

   Options parameters: none

12.5.  SDP Attributes

   This document registers encryption is an
   important element in the following new SDP attributes.

12.5.1.  'record' SDP Attribute

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

   Attribute name: record

   Long overall SIPREC solution; therefore SRC and
   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 attribute name: Recording Indication

   Type of attribute: session source authentication.  They
   do not contain or media level

   Subject require a specific keying mechanism.

   When RTP/SAVP or RTP/SAVPF is used, RS can choose to charset: no

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

   Allowed attribute values: on, off, paused

12.5.2.  '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 same or
   different security keys than the ones used in the CS.  Some SRCs are
   designed to simply replicate RTP packets from the CS media level

   Subject stream to charset: no

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

   Allowed attribute values: on, off, pause, nopreference

13. SRC will be reusing the same keys as the CS.  In
   this case, the SRC MUST secure the SDP with SDP Security Considerations

   The recording session is fundamentally a standard SIP dialog
   [RFC3261], therefore, Descriptions
   (SDES) [RFC4568] in the recording session can reuse any of RS with at least the
   existing SIP same level of security mechanism available for securing the recorded
   media as well
   as metadata.  Other the CS.  The risk of lowering the level of security considerations are
   outlined in the use cases and requirements document [RFC6341].

13.1.  RTP handling

   In many scenarios RS for
   this case is that it will be critical effectively become a downgrade attack on
   the CS since the same key is used for both CS and RS.

   For SRCs that perform transcoding or mixing of media before sending
   to the SRS, the SRC MUST negotiate a different security key than the
   one being used in the CS, to ensure that the media transported
   between security in the CS is
   not compromised by the SRC when reusing the same security key.

12.3.  Metadata

   Metadata contains sensitive information such as the address of record
   of the participants and SRS to be protected.  Media encryption other extension data placed by the SRC.  It
   is an
   important element essential to protect the content of the metadata in the overall SIPREC solution, therefore, it RS.  Since
   metadata is
   RECOMMENDED that SRC and SRS support RTP/SAVP [RFC3711] and RTP/SAVPF
   [RFC5124].  RTP/SAVP a content type transmitted in SIP signalling, metadata
   SHOULD be protected at the transport level by SIPS/TLS.

12.4.  Storage and RTP/SAVPF provide media encryption,
   integrity protection, replay protection, playback

   While storage and a limited form playback of source
   authentication.  They do not contain or require a specific keying
   mechanism.

13.2.  Authentication and Authorization

   The the call recording session reuses is beyond the SIP mechanism scope
   of this document, it is worthwhile to challenge requests mention here that it is based on HTTP authentication.  The mechanism relies on 401
   and 407 SIP responses as well as other SIP header fields also
   important for carrying
   challenges the recording storage and credentials.

   The SRS may have its own set playback to provide a level
   of recording policies security that is comparable to authorize
   recording requests from the SRC.  The use communication session.  It
   would defeat the purpose of securing both the communication session
   and the recording policies is
   outside session mentioned in the scope of previous sections if the Session Recording Protocol.

14.
   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.

15.

14.  References

15.1.

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-07 (work in progress),
              July 2012.

   [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.

   [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.

15.2.

14.2.  Informative References

   [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-05 draft-ietf-siprec-architecture-06
              (work in progress), May September 2012.

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

   [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.

   [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

   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