Internet DRAFT - draft-ietf-siprec-protocol
draft-ietf-siprec-protocol
SIPREC L. Portman
Internet-Draft NICE Systems
Intended status: Standards Track H. Lum, Ed.
Expires: March 28, 2016 Genesys
C. Eckel
Cisco
A. Johnston
Avaya
A. Hutton
Unify
September 25, 2015
Session Recording Protocol
draft-ietf-siprec-protocol-18
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. This document considers only
active recording, where the SRC purposefully streams media to an SRS
and all participating user agents are notified of the recording.
Passive recording, where a recording device detects media directly
from the network (e.g., using port-mirroring techniques), is outside
the scope of this document. In addition, lawful intercept is outside
the scope of this document.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
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This Internet-Draft will expire on March 28, 2016.
Copyright Notice
Copyright (c) 2015 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
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the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
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 . . . . . . . . . . . . . . 8
5.3. Receiving recording indications and providing recording
preferences . . . . . . . . . . . . . . . . . . . . . . . 9
6. SIP Handling . . . . . . . . . . . . . . . . . . . . . . . . 11
6.1. Procedures at the SRC . . . . . . . . . . . . . . . . . . 11
6.1.1. Initiating a Recording Session . . . . . . . . . . . 11
6.1.2. SIP extensions for recording indication and
preference . . . . . . . . . . . . . . . . . . . . . 11
6.2. Procedures at the SRS . . . . . . . . . . . . . . . . . . 12
6.3. Procedures for Recording-aware User Agents . . . . . . . 12
7. SDP Handling . . . . . . . . . . . . . . . . . . . . . . . . 13
7.1. Procedures at the SRC . . . . . . . . . . . . . . . . . . 13
7.1.1. SDP handling in RS . . . . . . . . . . . . . . . . . 13
7.1.1.1. Handling media stream updates . . . . . . . . . . 14
7.1.2. Recording indication in CS . . . . . . . . . . . . . 15
7.1.3. Recording preference in CS . . . . . . . . . . . . . 16
7.2. Procedures at the SRS . . . . . . . . . . . . . . . . . . 16
7.3. Procedures for Recording-aware User Agents . . . . . . . 18
7.3.1. Recording indication . . . . . . . . . . . . . . . . 18
7.3.2. Recording preference . . . . . . . . . . . . . . . . 19
8. RTP Handling . . . . . . . . . . . . . . . . . . . . . . . . 20
8.1. RTP Mechanisms . . . . . . . . . . . . . . . . . . . . . 20
8.1.1. RTCP . . . . . . . . . . . . . . . . . . . . . . . . 20
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8.1.2. RTP Profile . . . . . . . . . . . . . . . . . . . . . 21
8.1.3. SSRC . . . . . . . . . . . . . . . . . . . . . . . . 21
8.1.4. CSRC . . . . . . . . . . . . . . . . . . . . . . . . 22
8.1.5. SDES . . . . . . . . . . . . . . . . . . . . . . . . 22
8.1.5.1. CNAME . . . . . . . . . . . . . . . . . . . . . . 22
8.1.6. Keepalive . . . . . . . . . . . . . . . . . . . . . . 22
8.1.7. RTCP Feedback Messages . . . . . . . . . . . . . . . 23
8.1.7.1. Full Intra Request . . . . . . . . . . . . . . . 23
8.1.7.2. Picture Loss Indicator . . . . . . . . . . . . . 23
8.1.7.3. Temporary Maximum Media Stream Bit Rate Request . 24
8.1.8. Symmetric RTP/RTCP for Sending and Receiving . . . . 24
8.2. Roles . . . . . . . . . . . . . . . . . . . . . . . . . . 25
8.2.1. SRC acting as an RTP Translator . . . . . . . . . . . 26
8.2.1.1. Forwarding Translator . . . . . . . . . . . . . . 26
8.2.1.2. Transcoding Translator . . . . . . . . . . . . . 26
8.2.2. SRC acting as an RTP Mixer . . . . . . . . . . . . . 27
8.2.3. SRC acting as an RTP Endpoint . . . . . . . . . . . . 28
8.3. RTP Session Usage by SRC . . . . . . . . . . . . . . . . 28
8.3.1. SRC Using Multiple m-lines . . . . . . . . . . . . . 28
8.3.2. SRC Using Mixing . . . . . . . . . . . . . . . . . . 29
8.4. RTP Session Usage by SRS . . . . . . . . . . . . . . . . 30
9. Metadata . . . . . . . . . . . . . . . . . . . . . . . . . . 31
9.1. Procedures at the SRC . . . . . . . . . . . . . . . . . . 31
9.2. Procedures at the SRS . . . . . . . . . . . . . . . . . . 33
10. Persistent Recording . . . . . . . . . . . . . . . . . . . . 35
11. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 35
11.1. Registration of Option Tags . . . . . . . . . . . . . . 35
11.1.1. siprec Option Tag . . . . . . . . . . . . . . . . . 35
11.1.2. record-aware Option Tag . . . . . . . . . . . . . . 36
11.2. Registration of media feature tags . . . . . . . . . . . 36
11.2.1. src feature tag . . . . . . . . . . . . . . . . . . 36
11.2.2. srs feature tag . . . . . . . . . . . . . . . . . . 36
11.3. New Content-Disposition Parameter Registrations . . . . 37
11.4. Media Type Registration . . . . . . . . . . . . . . . . 37
11.5. SDP Attributes . . . . . . . . . . . . . . . . . . . . . 37
11.5.1. 'record' SDP Attribute . . . . . . . . . . . . . . . 37
11.5.2. 'recordpref' SDP Attribute . . . . . . . . . . . . . 38
12. Security Considerations . . . . . . . . . . . . . . . . . . . 38
12.1. Authentication and Authorization . . . . . . . . . . . . 39
12.2. RTP handling . . . . . . . . . . . . . . . . . . . . . . 39
12.3. Metadata . . . . . . . . . . . . . . . . . . . . . . . . 40
12.4. Storage and playback . . . . . . . . . . . . . . . . . . 40
13. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 41
14. References . . . . . . . . . . . . . . . . . . . . . . . . . 41
14.1. Normative References . . . . . . . . . . . . . . . . . . 41
14.2. Informative References . . . . . . . . . . . . . . . . . 42
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 44
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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 with the architecture
[RFC7245], 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.
This document considers only active recording, where the SRC
purposefully streams media to an SRS and all participating user
agents are notified of the recording. Passive recording, where a
recording device detects media directly from the network (e.g., using
port-mirroring techniques), is outside the scope of this document.
In addition, lawful intercept is outside the scope of this document,
in accordance with [RFC2804].
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 [RFC7245].
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.
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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 a non-SIP
mechanism
5. Overview of operations
This section is informative and provides a description of recording
operations.
Section 6 describes the SIP communication in a recording session
between an SRC and an SRS, and the procedures for recording-aware
user agents participating in a CS. Section 7 describes the SDP in a
recording session, and the procedures for recording indications and
recording preferences. Section 8 describes the RTP handling in a
recording session. Section 9 describes the mechanism to deliver
recording metadata from the SRC to the SRS.
As mentioned in the architecture document [RFC7245], 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 relay
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.
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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 example call flows shows an SRC establishing a
recording session towards an SRS. The first call flow illustrates
UA(A) acting as the SRC. The second illustrates a B2BUA acting as
the SRC. Note that the SRC can choose when to establish the
Recording Session independent of the Communication Session, even
though the following call flows suggest 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) 200 OK | |
|<----------------------| |
| | |
|(3)RS INVITE with SDP | |
|--------------------------------------------->|
| | (4) 200 OK with SDP |
|<---------------------------------------------|
|(5)CS RTP | |
|======================>| |
|<======================| |
|(6)RS RTP | |
|=============================================>|
|=============================================>|
| | |
|(7)CS BYE | |
|---------------------->| |
|(8)RS BYE | |
|--------------------------------------------->|
| | |
Figure 1: Basic recording call flow with UA as SRC
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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 2: Basic recording call flow with B2BUA as SRC
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.
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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 [RFC3311] 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 any time.
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 the SRS loses the current state
due to internal failure. The SRS may optionally attach a reason
along with the snapshot request. This request allows both 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.
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SRC SRS
| |
|(1) INVITE (metadata snapshot 1) |
|---------------------------------------------------->|
| (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 3: Delivering metadata via SIP UPDATE
5.3. Receiving recording indications and providing recording
preferences
The SRC is responsible to provide recording indications to the
participants in the CS. A recording-aware UA supports receiving
recording indications via the SDP attribute a=record, and it can
specify a recording preference in the CS by including the SDP
attribute a=recordpref. The recording attribute is a declaration by
the SRC in the CS to indicate whether recording is taking place. The
recording preference attribute is a declaration by the recording-
aware UA in the CS to indicate its recording preference. A UA that
does not want to be recorded may still be notified recording is
occurring for a number of reasons (e.g., it was not capable of
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indicating its preference, its preference was ignored, etc.) If this
occurs, the UA's only mechanism to avoid being recorded is to
terminate its participation in the session.
To illustrate how the attributes are used, if 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 4: Recording indication and recording preference
After the call is established and recording is in progress, UA (B)
later decides to change the recording preference to no recording and
sends a 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.
Note that UA (B) could have explicitly indicated a recording
preference in (2), the 200 OK for the original INVITE. Indicating a
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preference of no recording in an initial INVITE or an initial
response to an INVITE may reduce the chance of a user being recorded
in the first place.
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.
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 MAY include the '+sip.src' feature tag to
indicate the that it is an SRC.
Since SIP Caller Preferences extensions are optional to implement for
routing proxies, there is no guarantee that a recording session will
be routed to an SRC or SRS. A new options tag is introduced:
"siprec". As per [RFC3261], only an SRC or an SRS can accept this
option tag in a recording session. An SRC MUST include the "siprec"
option tag in the Require header when initiating a Recording Session
so that UA's which do not support the session recording protocol
extensions will simply reject the INVITE request with a 420 Bad
Extension.
When an SRC receives a new INVITE, the SRC MUST only consider the SIP
session as a recording session when both the '+sip.srs' feature tag
and 'siprec' option tag are included in the INVITE request.
6.1.2. SIP extensions for recording indication and preference
For the communication session, the SRC MUST provide recording
indications to all participants in the CS. A participant UA in a CS
can indicate that it is recording-aware by providing the "record-
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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 in-band,
having a signed participant contract in place, etc.
An SRC in the CS may also indicate itself as a session recording
client by including the '+sip.src' feature tag. A recording-aware
participant can learn that an SRC is in the CS, and can set the
recording preference for the CS with the new SDP a=recordpref
attribute described in the SDP Handling section below.
6.2. Procedures at the 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 SHOULD
include the '+sip.srs' feature tag to indicate that it is an SRS.
An SRS MUST include the "siprec" option tag in the Require header as
per [RFC3261] when initiating a Recording Session so that UA's which
do not support the session recording protocol extensions will simply
reject the INVITE request with a 420 Bad Extension.
6.3. Procedures for Recording-aware User Agents
A recording-aware user agent is a participant in the CS that supports
the SIP and SDP extensions for receiving recording indications and
for requesting recording preferences for the call. A recording-aware
UA MUST indicate that it can accept reporting of recording indication
provided by the SRC with a new option tag "record-aware" when
initiating or establishing a CS, meaning including the "record-aware"
tag in the Supported header in the initial INVITE request or
response.
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A recording-aware UA MUST provide a recording indication to the end
user through an appropriate user interface, indicating whether
recording is on, off, or paused for each medium. Appropriate user
interfaces may include real-time notification or previously
established agreements that use of the device is subject to
recording. Some user agents that are automatons (e.g., IVR, media
server, PSTN gateway) may not have a user interface to render
recording indication. When such a user agent indicates recording
awareness, the UA SHOULD render recording indication through other
means, such as passing an in-band tone on the PSTN gateway, putting
the recording indication in a log file, or raising an application
event in a VoiceXML dialog. These 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 in-band.
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 a=label attribute only applies to the SDP and Metadata
conveyed in the bodies of the SIP request or response that the label
appeared in. Note that a recorded stream is distinct from a CS
stream; the metadata provides a list of participants that contribute
to each recorded stream.
The following is an example SDP offer from an SRC with both audio and
video recorded streams. Note that the following example contains
unfolded lines longer than 72 characters. These are captured between
<allOneLine> tags.
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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 5: Sample SDP offer from SRC with audio and video streams
7.1.1.1. Handling media stream updates
Over the lifetime of a recording session, the SRC can add and remove
recorded streams from the recording session for various reasons. For
example, when a CS stream is added or removed from the CS, or when a
CS is created or terminated if a recording session handles multiple
CSes. To remove a recorded stream from the recording session, the
SRC sends a new SDP offer where the port of the media stream to be
removed is set to zero, according to the procedures in [RFC3264]. To
add a recorded stream to the recording session, the SRC sends a new
SDP offer by adding a new media stream description or by reusing an
old media stream which had been previously disabled, according to the
procedures in [RFC3264].
The SRC can temporarily discontinue streaming and collection of
recorded media from the SRC to the SRS for reasons such as masking
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the recording. In this case, the SRC sends a new SDP offer and sets
the media stream to inactive (a=inactive) for each recorded stream to
be paused, as per the procedures in [RFC3264]. To resume streaming
and collection of recorded media, the SRC sends a new SDP offer and
sets the media stream to sendonly (a=sendonly). Note that a CS
itself may change the media stream direction by updating the SDP, for
example, by setting a=inactive for SDP hold. Media stream direction
changes in CS are conveyed in the metadata by the SRC. When a CS
media stream is changed to/from inactive, the effect on the
corresponding RS media stream is governed by SRC policy. The SRC MAY
have a local policy to pause an RS media stream when the
corresponding CS media stream is inactive, or it MAY leave the RS
media stream as sendonly.
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
CS media streams such as playing an in-band tone or an announcement
to the participants. A new 'record' SDP attribute is introduced to
allow the SRC to indicate recording state to a recording-aware UA in
a CS.
The 'record' SDP attribute appears at the media-level or session-
level in either SDP offer or answer. When the attribute is applied
at the session-level, the indication applies to all media streams in
the SDP. When the attribute is applied at the media-level, the
indication applies to the media stream only, and that overrides the
indication if also set at the session-level. Whenever the recording
indication needs to change, such as termination of recording, then
the SRC MUST initiate a reINVITE or UPDATE to update the SDP a=record
attribute.
The following is the ABNF of the 'record' attribute:
attribute =/ record-attr
; attribute defined in RFC 4566
record-attr = "record:" indication
indication = "on" / "off" / "paused"
on: Recording is in progress.
off: No recording is in progress.
paused: Recording is in progress but media is paused.
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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. If the SRC makes a change in recording state, the
SRC MUST report the new recording state in the a=record attribute in
the SDP answer or in a subsequent SDP offer.
7.2. Procedures at the SRS
Typically the SRS only receives RTP streams from the SRC; therefore,
the SDP offer/answer from the SRS normally sets each media stream to
receive media, by setting them with the a=recvonly attribute,
according to the procedures of [RFC3264]. When the SRS is not ready
to receive a recorded stream, the SRS sets the media stream as
inactive in the SDP offer or answer by setting it with an 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 an a=recvonly attribute.
The following is an example of an SDP answer from the 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.
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v=0
o=SRS 0 0 IN IP4 198.51.100.20
s=-
c=IN IP4 198.51.100.20
t=0 0
m=audio 10000 RTP/AVP 0
a=recvonly
a=label:1
m=video 10002 RTP/AVP 98
a=rtpmap:98 H264/90000
<allOneLine>
a=fmtp:98 profile-level-id=42A01E;
sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==
</allOneLine>
a=recvonly
a=label:2
m=audio 10004 RTP/AVP 0
a=recvonly
a=label:3
m=video 10006 RTP/AVP 98
a=rtpmap:98 H264/90000
<allOneLine>
a=fmtp:98 profile-level-id=42A01E;
sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==
</allOneLine>
a=recvonly
a=label:4
Figure 6: 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 MUST NOT add recorded streams in the recording session when
the SRS sends a new SDP offer. Similarly, when the SRS starts a
recording session, the SRS MUST initiate the INVITE without an SDP
offer to let the SRC generate the SDP offer with the streams to be
recorded.
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.
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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 7: SRS responding to offer with a=inactive
7.3. Procedures for Recording-aware User Agents
7.3.1. Recording indication
When a recording-aware UA receives an SDP offer or answer that
includes the a=record attribute, the UA provides an indication to the
end user whether the recording is on, off, or paused for each medium
based on the most recently received a=record SDP attribute for that
medium.
When a CS is traversed through multiple UAs such as a B2BUA or a
conference focus, each UA involved in the CS that is aware that the
CS is being recorded MUST provide the recording indication through
the a=record attribute to all other parties in the CS.
It is possible that more than one SRC is in the call path of the same
CS, but the recording indication attribute does not provide any hint
as to which SRC or how many SRCs are recording. An endpoint knows
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only that the call is being recorded. Furthermore, this attribute is
not used as a request for a specific SRC to start/stop recording.
7.3.2. Recording preference
A participant in a CS MAY set the recording preference in the CS to
be recorded or not recorded at session establishment or during the
session. A new 'recordpref' SDP attribute is introduced, and the
participant in CS may set this recording preference attribute in any
SDP offer/answer at session establishment time or during the session.
The SRC is not required to honor the recording preference from a
participant based on local policies at the SRC, and the participant
can learn the recording indication through the a=record SDP attribute
as described in the above section.
The SDP a=recordpref attribute can appear at the media-level or
session-level and can appear in an SDP offer or answer. When the
attribute is applied at the session-level, the recording preference
applies to all media stream in the SDP. When the attribute is
applied at the media-level, the recording preference applies to the
media stream only, and that overrides the recording preference if
also set at the session-level. The user agent can change the
recording preference by changing the a=recordpref attribute in
subsequent SDP offer or answer. The absence of the a=recordpref
attribute in the SDP indicates that the UA has no recording
preference.
The following is the ABNF of the recordpref attribute:
attribute =/ recordpref-attr
; attribute defined in RFC 4566
recordpref-attr = "a=recordpref:" pref
pref = "on" / "off" / "pause" / "nopreference"
on: Sets the preference to record if it has not already been
started. If the recording is currently paused, the preference is
to resume recording.
off: Sets the preference for no recording. If recording has already
been started, then the preference is to stop the recording.
pause: If the recording is currently in progress, sets the
preference to pause the recording.
nopreference: To indicate that the UA has no preference on
recording.
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8. RTP Handling
This section provides recommendations and guidelines for RTP and RTCP
in the context of SIPREC. In order to communicate most effectively,
the Session Recording Client (SRC), the Session Recording Server
(SRS), and any Recording-aware User Agents (UAs) should utilize the
mechanisms provided by RTP in a well-defined and predicable manner.
It is the goal of this document to make the reader aware of these
mechanisms and provide recommendations and guidelines.
8.1. RTP Mechanisms
This section briefly describes important RTP/RTCP constructs and
mechanisms that are particularly useful within the context of SIPREC.
8.1.1. RTCP
The RTP data transport is augmented by a control protocol (RTCP) to
allow monitoring of the data delivery. RTCP, as defined in
[RFC3550], is based on the periodic transmission of control packets
to all participants in the RTP session, using the same distribution
mechanism as the data packets. Support for RTCP is REQUIRED, per
[RFC3550], and it provides, among other things, the following
important functionality in relation to SIPREC:
1) Feedback on the quality of the data distribution
This feedback from the receivers may be used to diagnose faults in
the distribution. As such, RTCP is a well-defined and efficient
mechanism for the SRS to inform the SRC, and for the SRC to inform
Recording-aware UAs, of issues that arise with respect to the
reception of media that is to be recorded.
2) Carries a persistent transport-level identifier for an RTP source
called the canonical name or CNAME
The SSRC identifier may change if a conflict is discovered or a
program is restarted, in which case receivers can use the CNAME to
keep track of each participant. Receivers may also use the CNAME to
associate multiple data streams from a given participant in a set of
related RTP sessions, for example to synchronize audio and video.
Synchronization of media streams is also facilitated by the NTP and
RTP timestamps included in RTCP packets by data senders.
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8.1.2. RTP Profile
The RECOMMENDED RTP profiles for the SRC, SRS, and Recording-aware
UAs are "Extended Secure RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/SAVPF)" [RFC5124], when using
encrypted RTP streams, and "Extended RTP Profile for Real-time
Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)"
[RFC4585], when using non-encrypted media streams. However, as these
are not requirements, some implementations may use "The Secure Real-
time Transport Protocol (SRTP)" [RFC3711], and "RTP Profile for Audio
and Video Conferences with Minimal Control" [RFC3551]. Therefore, it
is RECOMMENDED that the SRC, SRS, and Recording-aware UAs not rely
entirely on RTP/SAVPF or RTP/AVPF for core functionality that may be
at least partially achievable using RTP/SAVP and RTP/AVP.
AVPF and SAVPF provide an improved RTCP timer model that allows more
flexible transmission of RTCP packets in response to events, rather
than strictly according to bandwidth. AVPF-based codec control
messages provide efficient mechanisms for an SRC, SRS, and Recording-
aware UAs to handle events such as scene changes, error recovery, and
dynamic bandwidth adjustments. These messages are discussed in more
detail later in this document.
SAVP and SAVPF provide media encryption, integrity protection, replay
protection, and a limited form of source authentication. They do not
contain or require a specific keying mechanism.
8.1.3. SSRC
The synchronization source (SSRC), as defined in [RFC3550], is
carried in the RTP header and in various fields of RTCP packets. It
is a random 32-bit number that is required to be globally unique
within an RTP session. It is crucial that the number be chosen with
care in order that participants on the same network or starting at
the same time are not likely to choose the same number. Guidelines
regarding SSRC value selection and conflict resolution are provided
in [RFC3550].
The SSRC may also be used to separate different sources of media
within a single RTP session. For this reason as well as for conflict
resolution, it is important that the SRC, SRS, and Recording-aware
UAs handle changes in SSRC values and properly identify the reason of
the change. The CNAME values carried in RTCP facilitate this
identification.
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8.1.4. CSRC
The contributing source (CSRC), as defined in [RFC3550], identifies
the source of a stream of RTP packets that has contributed to the
combined stream produced by an RTP mixer. The mixer inserts a list
of the SSRC identifiers of the sources that contributed to the
generation of a particular packet into the RTP header of that packet.
This list is called the CSRC list. It is RECOMMENDED that an SRC or
Recording-aware UA, when acting as a mixer, set the CSRC list
accordingly, and that the SRC and SRS interpret the CSRC list per
[RFC3550] when received.
8.1.5. SDES
The Source Description (SDES), as defined in [RFC3550], contains an
SSRC/CSRC identifier followed by a list of zero or more items, which
carry information about the SSRC/CSRC. End systems send one SDES
packet containing their own source identifier (the same as the SSRC
in the fixed RTP header). A mixer sends one SDES packet containing a
chunk for each contributing source from which it is receiving SDES
information, or multiple complete SDES packets if there are more than
31 such sources.
The ability to identify individual contributing sources is important
in the context of SIPREC. Metadata [I-D.ietf-siprec-metadata]
provides a mechanism to achieve this at the signaling level. SDES
provides a mechanism at the RTP level.
8.1.5.1. CNAME
The Canonical End-Point Identifier (CNAME), as defined in [RFC3550],
provides the binding from the SSRC identifier to an identifier for
the source (sender or receiver) that remains constant. It is
important the SRC and Recording-aware UAs generate CNAMEs
appropriately and that the SRC and SRS interpret and use them for
this purpose. Guidelines for generating CNAME values are provided in
"Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names
(CNAMEs)" [RFC7022].
8.1.6. Keepalive
It is anticipated that media streams in SIPREC may exist in an
inactive state for extended periods of times for any of a number of
valid reasons. In order for the bindings and any pinholes in NATs/
firewalls to remain active during such intervals, it is RECOMMENDED
that the SRC, SRS, and Recording-aware UAs follow the keep-alive
procedure recommended in "Application Mechanism for Keeping Alive the
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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 in the RTP Audio-Visual Profile with Feedback
(AVPF)" [RFC5104] specifies extensions to the messages defined in
AVPF [RFC4585]. Support for and proper usage of these messages is
important to SRC, SRS, and Recording-aware UA implementations. Note
that these messages are applicable only when using the AVPF or SAVPF
RTP profiles
8.1.7.1. Full Intra Request
A Full Intra Request (FIR) Command, when received by the designated
media sender, requires that the media sender sends a Decoder Refresh
Point at the earliest opportunity. Using a decoder refresh point
implies refraining from using any picture sent prior to that point as
a reference for the encoding process of any subsequent picture sent
in the stream.
Decoder refresh points, especially Intra or IDR pictures for H.264
video codecs, are in general several times larger in size than
predicted pictures. Thus, in scenarios in which the available bit
rate is small, the use of a decoder refresh point implies a delay
that is significantly longer than the typical picture duration.
8.1.7.1.1. SIP INFO for FIR
"XML Schema for Media Control" [RFC5168] defines an Extensible Markup
Language (XML) Schema for video fast update. Implementations are
discouraged from using the method described except for backward
compatibility purposes. Implementations SHOULD use FIR messages
instead.
To make sure a common mechanism exists between the SRC and SRS, the
SRS MUST support both mechanisms (FIR and SIP INFO), using FIR when
negotiated successfully with the SRC, and using SIP INFO otherwise.
8.1.7.2. Picture Loss Indicator
Picture Loss Indication (PLI), as defined in [RFC4585], informs the
encoder of the loss of an undefined amount of coded video data
belonging to one or more pictures. [RFC4585] recommends using PLI
instead of FIR to recover from errors. FIR is appropriate only in
situations where not sending a decoder refresh point would render the
video unusable for the users. Examples where sending FIR is
appropriate include a multipoint conference when a new user joins the
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conference and no regular decoder refresh point interval is
established, and a video switching MCU that changes streams.
Appropriate use of PLI and FIR is important to ensure with minimum
overhead that the recorded video is usable (e.g., the necessary
reference frames exist for a player to render the recorded video).
8.1.7.3. Temporary Maximum Media Stream Bit Rate Request
A receiver, translator, or mixer uses the Temporary Maximum Media
Stream Bit Rate Request (TMMBR) to request a sender to limit the
maximum bit rate for a media stream to the provided value.
Appropriate use of TMMBR facilitates rapid adaptation to changes in
available bandwidth.
8.1.7.3.1. Renegotiation of SDP bandwidth attribute
If it is likely that the new value indicated by TMMBR will be valid
for the remainder of the session, the TMMBR sender is expected to
perform a renegotiation of the session upper limit using the session
signaling protocol. Therefore for SIPREC, implementations are
RECOMMENDED to use TMMBR for temporary changes, and renegotiation of
bandwidth via SDP offer/answer for more permanent changes.
8.1.8. Symmetric RTP/RTCP for Sending and Receiving
Within an SDP offer/answer exchange, RTP entities choose the RTP and
RTCP transport addresses (i.e., IP addresses and port numbers) on
which to receive packets. When sending packets, the RTP entities may
use the same source port or a different source port as those signaled
for receiving packets. When the transport address used to send and
receive RTP is the same, it is termed "symmetric RTP" [RFC4961].
Likewise, when the transport address used to send and receive RTCP is
the same, it is termed "symmetric RTCP" [RFC4961].
When sending RTP, it is REQUIRED to use symmetric RTP. When sending
RTCP, it is REQUIRED to use symmetric RTCP. Although an SRS will not
normally send RTP, it will send RTCP as well as receive RTP and RTCP.
Likewise, although an SRC will not normally receive RTP from the SRS,
it will receive RTCP as well as send RTP and RTCP.
Note: Symmetric RTP and symmetric RTCP are different from RTP/RTCP
multiplexing [RFC5761].
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8.2. Roles
An SRC has the task of gathering media from the various UAs in one or
more Communication Sessions (CSs) and forwarding the information to
the SRS within the context of a corresponding Recording Session (RS).
There are numerous ways in which an SRC may do this, 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 |
| | +-- Metadata -->| |
| | | +---------+
v v |
+---------+
| SRC |
|---------| (Communication Session) +---------+
| |<----------SIP---------->| |
| UA-A | | UA-B |
| |<-------RTP/RTCP-------->| |
+---------+ +---------+
Figure 8: 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 9: B2BUA as SRC
The following subsections define a set of roles an SRC may choose to
play based on its position with respect to a UA within a CS, and an
SRS within an RS. A CS and a corresponding RS are independent
sessions; therefore, an SRC may play a different role within a CS
than it does within the corresponding RS.
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8.2.1. SRC acting as an RTP Translator
The SRC may act as a translator, as defined in [RFC3550]. A defining
characteristic of a translator is that it forwards RTP packets with
their SSRC identifier intact. There are two types of translators,
one that simply forwards, and another that performs transcoding
(e.g., from one codec to another) in addition to forwarding.
8.2.1.1. Forwarding Translator
When acting as a forwarding translator, RTP received as separate
streams from different sources (e.g., from different UAs with
different SSRCs) cannot be mixed by the SRC and MUST be sent
separately to the SRS. All RTCP reports MUST be passed by the SRC
between the UAs and the SRS, such that the UAs and SRS are able to
detect any SSRC collisions.
RTCP Sender Reports generated by a UA sending a stream MUST be
forwarded to the SRS. RTCP Receiver Reports generated by the SRS
MUST be forwarded to the relevant UA.
UAs may receive multiple sets of RTCP Receiver Reports, one or more
from other UAs participating in the CS, and one from the SRS
participating in the RS. A UA SHOULD process the RTCP Receiver
Reports from the SRS if it is recording-aware.
If SRTP is used on both the CS and the RS, decryption and/or re-
encryption may occur. For example, if different keys are used, it
will occur. If the same keys are used, it need not occur.
Section 12 provides additional information on SRTP and keying
mechanisms.
If packet loss occurs, either from the UA to the SRC or from the SRC
to the SRS, the SRS SHOULD detect and attempt to recover from the
loss. The SRC does not play a role in this other than forwarding the
associated RTP and RTCP packets.
8.2.1.2. Transcoding Translator
When acting as a transcoding translator, an SRC MAY perform
transcoding (e.g., from one codec to another), and this may result in
a different rate of packets between what the SRC receives on the CS
and what the SRC sends on the RS. As when acting as a forwarding
translator, RTP received as separate streams from different sources
(e.g., from different UAs with different SSRCs) cannot be mixed by
the SRC and MUST be sent separately to the SRS. All RTCP reports
MUST be passed by the SRC between the UAs and the SRS, such that the
UAs and SRS are able to detect any SSRC collisions.
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RTCP Sender Reports generated by a UA sending a stream MUST be
forwarded to the SRS. RTCP Receiver Reports generated by the SRS
MUST be forwarded to the relevant UA. The SRC may need to manipulate
the RTCP Receiver Reports to take account of any transcoding that has
taken place.
UAs may receive multiple sets of RTCP Receiver Reports, one or more
from other UAs participating in the CS, and one from the SRS
participating in the RS. A Recording-aware UA SHOULD be prepared to
process the RTCP Receiver Reports from the SRS, whereas a recording
unaware UA may discard such RTCP packets as not of relevance.
If SRTP is used on both the CS and the RS, decryption and/or re-
encryption may occur. For example, if different keys are used, it
will occur. If the same keys are used, it need not occur.
Section 12 provides additional information on SRTP and keying
mechanisms.
If packet loss occurs, either from the UA to the SRC or from the SRC
to the SRS, the SRS SHOULD detect and attempt to recover from the
loss. The SRC does not play a role in this other than forwarding the
associated RTP and RTCP packets.
8.2.2. SRC acting as an RTP Mixer
In the case of the SRC acting as a RTP mixer, as defined in
[RFC3550], the SRC combines RTP streams from different UAs and sends
them towards the SRS using its own SSRC. The SSRCs from the
contributing UA SHOULD be conveyed as CSRCs identifiers within this
stream. The SRC may make timing adjustments among the received
streams and generate its own timing on the stream sent to the SRS.
Optionally an SRC acting as a mixer can perform transcoding, and can
even cope with different codings received from different UAs. RTCP
Sender Reports and Receiver Reports are not forwarded by an SRC
acting as mixer, but there are requirements for forwarding RTCP
Source Description (SDES) packets. The SRC generates its own RTCP
Sender and Receiver reports toward the associated UAs and SRS.
The use of SRTP between the SRC and the SRS for the RS is independent
of the use of SRTP between the UAs and SRC for the CS. Section 12
provides additional information on SRTP and keying mechanisms.
If packet loss occurs from the UA to the SRC, the SRC SHOULD detect
and attempt to recover from the loss. If packet loss occurs from the
SRC to the SRS, the SRS SHOULD detect and attempt to recover from the
loss.
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8.2.3. SRC acting as an RTP Endpoint
The case of the SRC acting as an RTP endpoint, as defined in
[RFC3550], is similar to the mixer case, except that the RTP session
between the SRC and the SRS is considered completely independent from
the RTP session that is part of the CS. The SRC can, but need not,
mix RTP streams from different participants prior to sending to the
SRS. RTCP between the SRC and the SRS is completely independent of
RTCP on the CS.
The use of SRTP between the SRC and the SRS for the RS is independent
of the use of SRTP between the UAs and SRC for the CS. Section 12
provides additional information on SRTP and keying mechanisms.
If packet loss occurs from the UA to the SRC, the SRC SHOULD detect
and attempt to recover from the loss. If packet loss occurs from the
SRC to the SRS, the SRS SHOULD detect and attempt to recover from the
loss.
8.3. RTP Session Usage by SRC
There are multiple ways that an SRC may choose to deliver recorded
media to an SRS. In some cases, it may use a single RTP session for
all media within the RS, whereas in others it may use multiple RTP
sessions. The following subsections provide examples of basic RTP
session usage by the SRC, including a discussion of how the RTP
constructs and mechanisms covered previously are used. An SRC may
choose to use one or more of the RTP session usages within a single
RS. For the purpose of base interoperability between SRC and SRS, an
SRC MUST support separate m-lines in SDP, one per CS media direction.
The set of RTP session usages described is not meant to be
exhaustive.
8.3.1. SRC Using Multiple m-lines
When using multiple m-lines, an SRC includes each m-line in an SDP
offer to the SRS. The SDP answer from the SRS MUST include all
m-lines, with any rejected m-lines indicated with a zero port, per
[RFC3264]. Having received the answer, the SRC starts sending media
to the SRS as indicated in the answer. Alternatively, if the SRC
deems the level of support indicated in the answer to be
unacceptable, it may initiate another SDP offer/answer exchange in
which an alternative RTP session usage is negotiated.
In order to preserve the mapping of media to participant within the
CSs in the RS, the SRC SHOULD map each unique CNAME within the CSs to
a unique CNAME within the RS. Additionally, the SRC SHOULD map each
unique combination of CNAME/SSRC within the CSs to a unique CNAME/
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SSRC within the RS. In doing so, the SRC may act as an RTP
translator or as an RTP endpoint.
The following figure illustrates a case in which each UA represents a
participant contributing two RTP sessions (e.g., one for audio and
one for video), each with a single SSRC. The SRC acts as an RTP
translator and delivers the media to the SRS using four RTP sessions,
each with a single SSRC. The CNAME and SSRC values used by the UAs
within their media streams are preserved in the media streams from
the SRC to the SRS.
+---------+
+------------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 10: SRC Using Multiple m-lines
8.3.2. SRC Using Mixing
When using mixing, the SRC combines RTP streams from different
participants and sends them towards the SRS using its own SSRC. The
SSRCs from the contributing participants SHOULD be conveyed as CSRCs
identifiers. The SRC includes one m-line for each RTP session in an
SDP offer to the SRS. The SDP answer from the SRS MUST include all
m-lines, with any rejected m-lines indicated with the zero port, per
[RFC3264]. Having received the answer, the SRC starts sending media
to the SRS as indicated in the answer.
In order to preserve the mapping of media to participant within the
CSs in the RS, the SRC SHOULD map each unique CNAME within the CSs to
a unique CNAME within the RS. Additionally, the SRC SHOULD map each
unique combination of CNAME/SSRC within the CSs to a unique CNAME/
SSRC within the RS. The SRC MUST avoid SSRC collisions, rewriting
SSRCs if necessary when used as CSRCs in the RS. In doing so, the
SRC acts as an RTP mixer.
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In the event the SRS does not support this usage of CSRC values, it
relies entirely on the SIPREC metadata to determine the participants
included within each mixed stream.
The following figure illustrates a case in which each UA represents a
participant contributing two RTP sessions (e.g., one for audio and
one for video), each with a single SSRC. The SRC acts as an RTP
mixer and delivers the media to the SRS using two RTP sessions,
mixing media from each participant into a single RTP session
containing a single SSRC and two CSRCs.
SSRC Sa +---------+
+-------CSRC Aa,Ba--->| |
| | |
| SSRC Sv | SRS |
| +---CSRC Av,Bv--->| |
| | +---------+
| |
+----------+
+---------+ | SRC | +---------+
| |---SSRC Aa-->|(CNAME-S, |<--SSRC Ba---| |
| UA-A | | CNAME-A, | | UA-B |
|(CNAME-A)|---SSRC Av-->| CNAME-B) |<--SSRC Bv---|(CNAME-B)|
+---------+ +----------+ +---------+
Figure 11: SRC Using Mixing
8.4. RTP Session Usage by SRS
An SRS that supports recording an audio CS MUST support SRC usage of
separate audio m-lines in SDP, one per CS media direction. An SRS
that supports recording a video CS MUST support SRC usage of separate
video m-lines in SDP, one per CS media direction. Therefore, for an
SRS supporting a typical audio call, the SRS has to support receiving
at least two audio m-lines. For an SRS supporting a typical audio
and video call, the SRS has to support receiving at least four total
m-lines in the SDP, two audio m-lines and two video m-lines.
These requirements allow an SRS to be implemented that supports video
only, without requiring support for audio recording. They also allow
an SRS to be implemented that supports recording only one direction
of one stream in a CS; for example, an SRS designed to record
security monitoring cameras that only send (not receive) video
without any audio. These requirements were not written to prevent
other modes being implemented and used, such as using a single m-line
and mixing the separate audio streams together. Rather, the
requirements were written to provide a common base mode to implement
for the sake of interoperability. It is important to note that an
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SRS implementation supporting the common base may not record all
media streams in a CS if a participant supports more than one m-line
in a video call, such as one for camera and one for presentation.
SRS implementations may support other modes as well, but have to at
least support the ones above such that they interoperate in the
common base mode for basic interoperability.
9. Metadata
Some metadata attributes are contained in SDP, and others are
contained in a new content type "application/rs-metadata". The
format of the metadata is described as part of the mechanism in
[I-D.ietf-siprec-metadata]. A new "disposition-type" of Content-
Disposition is defined for the purpose of carrying metadata. The
value is "recording-session", which indicates the "application/rs-
metadata" content contains metadata to be handled by the SRS.
9.1. Procedures at the SRC
The SRC MUST send metadata to the SRS in an RS. The SRC SHOULD send
metadata as soon as it becomes available and whenever it changes.
Cases in which an SRC may be justified in waiting temporarily before
sending metadata include:
o waiting for a previous metadata exchange to complete (i.e., the
SRC cannot send another SDP offer until the previous offer/answer
completes, and may prefer not to send an UPDATE during this time
either).
o constraining the signaling rate on the RS.
o sending metadata when key events occur rather than for every event
that has any impact on metadata.
The SRC may also be configured to suppress certain metadata out of
concern for privacy or perceived lack of need for it to be included
in the recording.
Metadata sent by the SRC is categorized as either a full metadata
snapshot or a partial update. A full metadata snapshot describes all
metadata associated with the RS. The SRC MAY send a full metadata
snapshot at any time. The SRC MAY send a partial update only if a
full metadata snapshot has been sent previously.
The SRC MAY send metadata (either a full metadata snapshot or a
partial update) in an INVITE request, an UPDATE request [RFC3311], or
a 200 response to an offerless INVITE from the SRS. If the metadata
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contains a reference to any SDP labels, the request containing the
metadata MUST also contain an SDP offer that defines those labels.
When a SIP message contains both an SDP offer and metadata, the
request body MUST have content type "multipart/mixed", with one
subordinate body part containing the SDP offer and another containing
the metadata. When a SIP message contains only an SDP offer or
metadata, the "multipart/mixed" container is optional.
The SRC SHOULD include a full metadata snapshot in the initial INVITE
request establishing the RS. If metadata is not yet available (e.g.,
an RS established in absence of a CS), the SRC SHOULD send a full
metadata snapshot as soon as metadata becomes available.
If the SRC receives a snapshot request from the SRS, it MUST
immediately send a full metadata snapshot.
The following is an example of a full metadata snapshot sent by the
SRC in the initial INVITE request:
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INVITE sip:recorder@example.com SIP/2.0
Via: SIP/2.0/TCP src.example.com;branch=z9hG4bKdf6b622b648d9
From: <sip:2000@example.com>;tag=35e195d2-947d-4585-946f-09839247
To: <sip:recorder@example.com>
Call-ID: d253c800-b0d1ea39-4a7dd-3f0e20a
CSeq: 101 INVITE
Max-Forwards: 70
Require: siprec
Accept: application/sdp, application/rs-metadata
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 for the recording session
9.2. Procedures at the SRS
The SRS receives metadata updates from the SRC in INVITE and UPDATE
requests. Since the SRC can send partial updates based on the
previous update, the SRS needs to keep track of the sequence of
updates from the SRC.
In the case of an internal failure at the SRS, the SRS may fail to
recognize a partial update from the SRC. The SRS may be able to
recover from the internal failure by requesting a full metadata
snapshot from the SRC. Certain errors, such as syntax errors or
semantic errors in the metadata information, are likely caused by an
error on the SRC side, and it is likely the same error will occur
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again even when a full metadata snapshot is requested. In order to
avoid repeating the same error, the SRS can simply terminate the
recording session when a syntax error or semantic error is detected
in the metadata.
The SRS MAY explicitly request a full metadata snapshot by sending an
UPDATE request. This request MUST contain a body with content
disposition type "recording-session", and MUST NOT contain an SDP
body. The SRS MUST NOT request a full metadata snapshot in an UPDATE
response or in any other SIP transaction. The format of the content
is "application/rs-metadata", and the body is an XML document, the
format of which is defined in [I-D.ietf-siprec-metadata]. 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
Content-Length: [length]
<?xml version="1.0" encoding="UTF-8"?>
<requestsnapshot xmlns='urn:ietf:params:xml:ns:recording:1'>
<requestreason xml:lang="it">SRS internal error</requestreason>
</requestsnapshot>
Figure 13: Metadata Request
Note that UPDATE was chosen for the SRS to request metadata snapshot
because it can be sent regardless of the state of the dialog. This
was seen as better than requiring support for both UPDATE and re-
INVITE for this operation.
When the SRC receives a request for a metadata snapshot, it MUST
immediately provide a full metadata snapshot in a separate INVITE or
UPDATE transaction. Any subsequent partial updates will not be
dependent on any metadata sent prior to this full metadata snapshot.
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The metadata received by the SRS can contain ID elements used to
cross reference one element to another. An element containing the
definition of an ID, and an element containing a reference to that ID
will often be received from the same SRC. It is also valid for those
elements to be received from different SRCs, for example, when each
endpoint in the same CS act as an SRC to record the call and a common
ID refers to the same CS. The SRS MUST NOT consider this an error.
10. Persistent Recording
Persistent recording is a specific use case outlined in REQ-005 or
Use Case 4 in [RFC6341], where a recording session can be established
in the absence of a communication session. The SRC continuously
records media in a recording session to the SRS even in the absence
of a CS for all user agents that are part of persistent recording.
By allocating recorded streams and continuously sending recorded
media to the SRS, the SRC does not have to prepare new recorded
streams with a new SDP offer when a new communication session is
created and also does not impact the timing of the CS. The SRC only
needs to update the metadata when new communication sessions are
created.
When there is no communication session running on the devices with
persistent recording, there is no recorded media to stream from the
SRC to the SRS. In certain environments where Network Address
Translator (NAT) is used, typically a minimum of flow activity is
required to maintain the NAT binding for each port opened. Agents
that support Interactive Connectivity Establishment (ICE) solve this
problem. For non-ICE agents, in order not to lose the NAT bindings
for the RTP/RTCP ports opened for the recorded streams, the SRC and
SRS SHOULD follow the recommendations provided in [RFC6263] to
maintain the NAT bindings.
11. IANA Considerations
11.1. Registration of Option Tags
This specification registers two option tags. The required
information for this registration, as specified in [RFC3261], is as
follows.
11.1.1. siprec Option Tag
Name: siprec
Description: This option tag is for identifying that the SIP
session is for the purpose of a recording session. This is
typically not used in a Supported header. When present in a
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Require header in a request, it indicates that the UA is either an
SRC or SRS capable of handling a recording session.
11.1.2. record-aware Option Tag
Name: record-aware
Description: This option tag is to indicate the ability for the
user agent to receive recording indicators in media-level or
session-level SDP. When present in a Supported header, it
indicates that the UA can receive recording indicators in media-
level or session-level SDP.
11.2. Registration of media feature tags
This document registers two new media feature tags in the SIP tree
per the process defined in [RFC2506] and [RFC3840]
11.2.1. src feature tag
Media feature tag name: sip.src
ASN.1 Identifier: TBD at registration
Summary of the media feature indicated by this tag: This feature
tag indicates that the user agent is a Session Recording Client
for the purpose of a Recording Session.
Values appropriate for use with this feature tag: boolean
The feature tag is intended primarily for use in the following
applications, protocols, services, or negotiation mechanisms: This
feature tag is only useful for a Recording Session.
Examples of typical use: Routing the request to a Session
Recording Server.
Security Considerations: Security considerations for this media
feature tag are discussed in Section 11.1 of RFC 3840.
11.2.2. srs feature tag
Media feature tag name: sip.srs
ASN.1 Identifier: TBD at registration
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Summary of the media feature indicated by this tag: This feature
tag indicates that the user agent is a Session Recording Server
for the purpose of a Recording Session.
Values appropriate for use with this feature tag: boolean
The feature tag is intended primarily for use in the following
applications, protocols, services, or negotiation mechanisms: This
feature tag is only useful for a Recording Session.
Examples of typical use: Routing the request to a Session
Recording Client.
Security Considerations: Security considerations for this media
feature tag are discussed in Section 11.1 of RFC 3840.
11.3. New Content-Disposition Parameter Registrations
This document registers a new "disposition-type" value in Content-
Disposition header: recording-session.
recording-session: The body describes either:
* metadata about the recording session
* reason for metadata snapshot request
as determined by the MIME value indicated in the Content-Type.
11.4. Media Type Registration
11.5. SDP Attributes
This document registers the following new SDP attributes.
11.5.1. 'record' SDP Attribute
Contact names: Leon Portman leon.portman@gmail.com, Henry Lum
henry.lum@genesyslab.com
Attribute name: record
Long form attribute name: Recording Indication
Type of attribute: session or media-level
Subject to charset: no
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This attribute provides the recording indication for the session or
media stream.
Allowed attribute values: on, off, paused
11.5.2. 'recordpref' SDP Attribute
Contact names: Leon Portman leon.portman@nice.com, Henry Lum
henry.lum@genesyslab.com
Attribute name: recordpref
Long form attribute name: Recording Preference
Type of attribute: session or media-level
Subject to charset: no
This attribute provides the recording preference for the session or
media stream.
Allowed attribute values: on, off, pause, nopreference
12. Security Considerations
The recording session is fundamentally a standard SIP dialog
[RFC3261]; therefore, the recording session can reuse any of the
existing SIP security mechanisms available for securing the session
signaling, the recorded media, and the metadata. The use cases and
requirements document [RFC6341] outlines the general security
considerations, and this document describes specific security
recommendations.
The SRC and SRS MUST support SIP with TLS version 1.2, SHOULD follow
the best practices when using TLS as per [RFC7525], and MAY use SIPS
with TLS as per [RFC5630]. The Recording Session MUST be at least as
secure as the Communication Session, meaning using at least the same
strength of cipher suite as the CS if the CS is secured. For
example, if the CS uses SIPS for signaling and RTP/SAVP for media,
then the RS may not use SIP or plain RTP unless other equivalent
security measures are in effect, since doing so would mean an
effective security downgrade. Examples of other potentially
equivalent security mechanisms include mutually-authenticated TLS for
the RS signaling channel or an appropriately protected network path
for the RS media component.
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12.1. Authentication and Authorization
At the transport level, the recording session uses TLS authentication
to validate the authenticity of the SRC and SRS. The SRC and SRS
MUST implement TLS mutual authentication for establishing the
recording session. Whether the SRC/SRS chooses to use TLS mutual
authentication is a deployment decision. In deployments where a UA
acts as its own SRC, this requires the UA have its own certificate as
needed for TLS mutual authentication. In deployments where the SRC
and the SRS are in the same administrative domain and have some other
means of assuring authenticity, the SRC and SRS may choose not to
authenticate each other, or to have the SRC authenticate the SRS
only. In deployments where the SRS can be hosted on a different
administrative domain, it is important to perform mutual
authentication to ensure the authenticity of both the SRC and the SRS
before transmitting any recorded media. The risk of not
authenticating the SRS is that the recording may be sent to an entity
other than the intended SRS, allowing a sensitive call recording to
be received by an attacker. On the other hand, the risk of not
authenticating the SRC is that an SRS will accept calls from an
unknown SRC and allow potential forgery of call recordings.
There may be scenarios in which the signaling between the SRC and SRS
is not direct, e.g., a SIP proxy exists between the SRC and the SRS.
In such scenarios, each hop is subject to the TLS mutual
authentication constraint and transitive trust at each hop is
utilized. Additionally, an SRC or SRS may use other existing SIP
mechanisms available, including but not limited to, Digest
Authentication [RFC3261], Asserted Identity [RFC3325], and Connected
Identity [RFC4916].
The SRS may have its own set of recording policies to authorize
recording requests from the SRC. The use of recording policies is
outside the scope of the Session Recording Protocol.
12.2. RTP handling
In many scenarios it will be critical for the media transported
between the SRC and the SRS to be protected. Media encryption is an
important element in the overall SIPREC solution; therefore the SRC
and the SRS MUST support RTP/SAVP [RFC3711] and RTP/SAVPF [RFC5124].
RTP/SAVP and RTP/SAVPF provide media encryption, integrity
protection, replay protection, and a limited form of source
authentication. They do not contain or require a specific keying
mechanism. At a minimum, the SRC and SRS MUST support the SDP
Security Descriptions (SDES) key negotiation mechanism [RFC4568].
For cases in which DTLS-SRTP is used to encrypt a CS media stream, an
SRC may use SRTP Encrypted Key Transport (EKT)
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[I-D.ietf-avtcore-srtp-ekt] in order to use SRTP-SDES in the RS
without needing to re-encrypt the media.
Note: When using EKT in this manner, it is possible for
participants in the CS to send traffic that appears to be from
other participants and have this forwarded by the SRC to the SRS
within the RS. If this is a concern (e.g. the RS is intended for
audit or compliance purposes), EKT is not an appropriate choice.
When RTP/SAVP or RTP/SAVPF is used, an SRC can choose to use the same
or different keys in the RS than the ones used in the CS. Some SRCs
are designed to simply replicate RTP packets from a CS media stream
to the SRS, in which case the SRC will use the same key in the RS as
used in the CS. In this case, the SRC MUST secure the SDP containing
the keying material in the RS with at least the same level of
security as in the CS. The risk of lowering the level of security in
the RS is that it will effectively become a downgrade attack on the
CS since the same key is used for both CS and RS.
SRCs that decrypt an encrypted CS media stream and re-encrypt it when
sending it to the SRS MUST use a different key than what is used for
the CS media stream, to ensure that it is not possible for someone
who has the key for the CS media stream to access recorded data they
are not authorized to access. In order to maintain a comparable
level of security, the key used in the RS SHOULD of equivalent or
greater strength than that used in the CS.
12.3. Metadata
Metadata contains sensitive information such as the address of record
of the participants and other extension data placed by the SRC. It
is essential to protect the content of the metadata in the RS. Since
metadata is a content type transmitted in SIP signaling, metadata
SHOULD be protected at the transport level by SIPS/TLS.
12.4. Storage and playback
While storage and playback of the call recording is beyond the scope
of this document, it is worthwhile to mention here that it is also
important for the recording storage and playback to provide a level
of security that is comparable to the communication session. It
would defeat the purpose of securing both the communication session
and the recording session mentioned in the previous sections if the
recording can be easily played back with a simple, unsecured HTTP
interface without any form of authentication or authorization.
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13. Acknowledgements
We want to thank John Elwell, Paul Kyzivat, Partharsarathi R, Ram
Mohan R, Hadriel Kaplan, Adam Roach, Miguel Garcia, Thomas Stach,
Muthu Perumal, Dan Wing, and Magnus Westerlund for their valuable
comments and inputs to this document.
14. References
14.1. Normative References
[I-D.ietf-siprec-metadata]
R, R., Ravindran, P., and P. Kyzivat, "Session Initiation
Protocol (SIP) Recording Metadata", draft-ietf-siprec-
metadata-18 (work in progress), August 2015.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<http://www.rfc-editor.org/info/rfc2119>.
[RFC2506] Holtman, K., Mutz, A., and T. Hardie, "Media Feature Tag
Registration Procedure", BCP 31, RFC 2506,
DOI 10.17487/RFC2506, March 1999,
<http://www.rfc-editor.org/info/rfc2506>.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
DOI 10.17487/RFC3261, June 2002,
<http://www.rfc-editor.org/info/rfc3261>.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
DOI 10.17487/RFC3264, June 2002,
<http://www.rfc-editor.org/info/rfc3264>.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <http://www.rfc-editor.org/info/rfc3550>.
[RFC3840] Rosenberg, J., Schulzrinne, H., and P. Kyzivat,
"Indicating User Agent Capabilities in the Session
Initiation Protocol (SIP)", RFC 3840,
DOI 10.17487/RFC3840, August 2004,
<http://www.rfc-editor.org/info/rfc3840>.
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[RFC4574] Levin, O. and G. Camarillo, "The Session Description
Protocol (SDP) Label Attribute", RFC 4574,
DOI 10.17487/RFC4574, August 2006,
<http://www.rfc-editor.org/info/rfc4574>.
[RFC7245] Hutton, A., Ed., Portman, L., Ed., Jain, R., and K. Rehor,
"An Architecture for Media Recording Using the Session
Initiation Protocol", RFC 7245, DOI 10.17487/RFC7245, May
2014, <http://www.rfc-editor.org/info/rfc7245>.
14.2. Informative References
[I-D.ietf-avtcore-srtp-ekt]
Mattsson, J., McGrew, D., and D. Wing, "Encrypted Key
Transport for Secure RTP", draft-ietf-avtcore-srtp-ekt-03
(work in progress), October 2014.
[RFC2804] IAB and , "IETF Policy on Wiretapping", RFC 2804,
DOI 10.17487/RFC2804, May 2000,
<http://www.rfc-editor.org/info/rfc2804>.
[RFC3311] Rosenberg, J., "The Session Initiation Protocol (SIP)
UPDATE Method", RFC 3311, DOI 10.17487/RFC3311, October
2002, <http://www.rfc-editor.org/info/rfc3311>.
[RFC3325] Jennings, C., Peterson, J., and M. Watson, "Private
Extensions to the Session Initiation Protocol (SIP) for
Asserted Identity within Trusted Networks", RFC 3325,
DOI 10.17487/RFC3325, November 2002,
<http://www.rfc-editor.org/info/rfc3325>.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
DOI 10.17487/RFC3551, July 2003,
<http://www.rfc-editor.org/info/rfc3551>.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, DOI 10.17487/RFC3711, March 2004,
<http://www.rfc-editor.org/info/rfc3711>.
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Description Protocol (SDP) Security Descriptions for Media
Streams", RFC 4568, DOI 10.17487/RFC4568, July 2006,
<http://www.rfc-editor.org/info/rfc4568>.
Portman, et al. Expires March 28, 2016 [Page 42]
Internet-Draft Session Recording Protocol September 2015
[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,
DOI 10.17487/RFC4585, July 2006,
<http://www.rfc-editor.org/info/rfc4585>.
[RFC4916] Elwell, J., "Connected Identity in the Session Initiation
Protocol (SIP)", RFC 4916, DOI 10.17487/RFC4916, June
2007, <http://www.rfc-editor.org/info/rfc4916>.
[RFC4961] Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
BCP 131, RFC 4961, DOI 10.17487/RFC4961, July 2007,
<http://www.rfc-editor.org/info/rfc4961>.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104,
February 2008, <http://www.rfc-editor.org/info/rfc5104>.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February
2008, <http://www.rfc-editor.org/info/rfc5124>.
[RFC5168] Levin, O., Even, R., and P. Hagendorf, "XML Schema for
Media Control", RFC 5168, DOI 10.17487/RFC5168, March
2008, <http://www.rfc-editor.org/info/rfc5168>.
[RFC5630] Audet, F., "The Use of the SIPS URI Scheme in the Session
Initiation Protocol (SIP)", RFC 5630,
DOI 10.17487/RFC5630, October 2009,
<http://www.rfc-editor.org/info/rfc5630>.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761,
DOI 10.17487/RFC5761, April 2010,
<http://www.rfc-editor.org/info/rfc5761>.
[RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for
Keeping Alive the NAT Mappings Associated with RTP / RTP
Control Protocol (RTCP) Flows", RFC 6263,
DOI 10.17487/RFC6263, June 2011,
<http://www.rfc-editor.org/info/rfc6263>.
[RFC6341] Rehor, K., Ed., Portman, L., Ed., Hutton, A., and R. Jain,
"Use Cases and Requirements for SIP-Based Media Recording
(SIPREC)", RFC 6341, DOI 10.17487/RFC6341, August 2011,
<http://www.rfc-editor.org/info/rfc6341>.
Portman, et al. Expires March 28, 2016 [Page 43]
Internet-Draft Session Recording Protocol September 2015
[RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla,
"Guidelines for Choosing RTP Control Protocol (RTCP)
Canonical Names (CNAMEs)", RFC 7022, DOI 10.17487/RFC7022,
September 2013, <http://www.rfc-editor.org/info/rfc7022>.
[RFC7525] Sheffer, Y., Holz, R., and P. Saint-Andre,
"Recommendations for Secure Use of Transport Layer
Security (TLS) and Datagram Transport Layer Security
(DTLS)", BCP 195, RFC 7525, DOI 10.17487/RFC7525, May
2015, <http://www.rfc-editor.org/info/rfc7525>.
Authors' Addresses
Leon Portman
NICE Systems
22 Zarhin Street
P.O. Box 690
Ra'anana 4310602
Israel
Email: leon.portman@gmail.com
Henry Lum (editor)
Genesys
1380 Rodick Road, Suite 201
Markham, Ontario L3R4G5
Canada
Email: henry.lum@genesyslab.com
Charles Eckel
Cisco
170 West Tasman Drive
San Jose, CA 95134
United States
Email: eckelcu@cisco.com
Alan Johnston
Avaya
St. Louis, MO 63124
Email: alan.b.johnston@gmail.com
Portman, et al. Expires March 28, 2016 [Page 44]
Internet-Draft Session Recording Protocol September 2015
Andrew Hutton
Unify
Brickhill Street
Milton Keynes MK15 0DJ
United Kingdom
Email: andrew.hutton@unify.com
Portman, et al. Expires March 28, 2016 [Page 45]
