Guidelines for End-to-End Support of the RTP Control Protocol (RTCP) in Back-to-Back User Agents (B2BUAs)
RFC 8079
| Document | Type | RFC - Proposed Standard (February 2017) | |
|---|---|---|---|
| Authors | Lorenzo Miniero , Sergio Garcia Murillo , Victor Pascual | ||
| Last updated | 2018-12-20 | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
| Formats | |||
| Reviews |
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| Additional resources | Mailing list discussion | ||
| Stream | WG state | Submitted to IESG for Publication | |
| Document shepherd | Christer Holmberg | ||
| Shepherd write-up | Show Last changed 2015-05-12 | ||
| IESG | IESG state | RFC 8079 (Proposed Standard) | |
| Action Holders |
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| Consensus boilerplate | Yes | ||
| Telechat date | (None) | ||
| Responsible AD | Ben Campbell | ||
| Send notices to | (None) | ||
| IANA | IANA review state | Version Changed - Review Needed | |
| IANA action state | No IANA Actions |
RFC 8079
Internet Engineering Task Force (IETF) L. Miniero
Request for Comments: 8079 Meetecho
Category: Standards Track S. Garcia Murillo
ISSN: 2070-1721 Medooze
V. Pascual
Oracle
February 2017
Guidelines for End-to-End Support of the RTP Control Protocol (RTCP) in
Back-to-Back User Agents (B2BUAs)
Abstract
SIP Back-to-Back User Agents (B2BUAs) are often designed to also be
on the media path, rather than just to intercept signalling. This
means that B2BUAs often implement an RTP or RTP Control Protocol
(RTCP) stack as well, thus leading to separate multimedia sessions
that the B2BUA correlates and bridges together. If not disciplined,
this behaviour can severely impact the communication experience,
especially when statistics and feedback information contained in RTCP
messages get lost because of mismatches in the reported data.
This document defines the proper behaviour B2BUAs should follow when
acting on both the signalling plane and media plane in order to
preserve the end-to-end functionality of RTCP.
Status of This Memo
This is an Internet Standards Track document.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 7841.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/rfc8079.
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Copyright Notice
Copyright (c) 2017 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
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Signalling/Media Plane B2BUAs . . . . . . . . . . . . . . . . 4
3.1. Media Relay . . . . . . . . . . . . . . . . . . . . . . . 5
3.2. Media-Aware Relay . . . . . . . . . . . . . . . . . . . . 6
3.3. Media Terminator . . . . . . . . . . . . . . . . . . . . 11
4. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 12
5. Security Considerations . . . . . . . . . . . . . . . . . . . 12
6. References . . . . . . . . . . . . . . . . . . . . . . . . . 13
6.1. Normative References . . . . . . . . . . . . . . . . . . 13
6.2. Informative References . . . . . . . . . . . . . . . . . 14
Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . . 15
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 16
1. Introduction
Session Initiation Protocol (SIP) [RFC3261] Back-to-Back User Agents
(B2BUAs) are SIP entities that can act as a logical combination of
both a User Agent Server (UAS) and a User Agent Client (UAC). As
such, their behaviour is not always completely adherent to standards
and can lead to unexpected situations. [RFC7092] presents a taxonomy
of the most commonly deployed B2BUA implementations and describes how
they differ in terms of the functionality and features they provide.
Such components often do not only act on the signalling plane
(intercepting and possibly modifying SIP messages), but also on the
media plane. This means that, in order to receive and manage all RTP
and RTCP [RFC3550] packets in a session, these components also
manipulate the session descriptions [RFC4566] in the related offer/
answer exchanges [RFC3264]. The reasons for such behaviour can be
different. The B2BUA may want, for instance, to provide transcoding
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functionality for participants with incompatible codecs, or it may
need the traffic to be directly handled for different reasons. This
can lead to several different topologies for RTP-based communication,
as documented in [RFC7667].
Whatever the reason, such behaviour does not come without a cost. In
fact, whenever a media-aware component is placed on the path between
two or more participants that want to communicate by means of RTP/
RTCP, the end-to-end nature of such protocols is broken. While this
may not be a problem for RTP packets, which can be quite easily
relayed, it definitely can cause serious issue for RTCP messages,
which carry important information and feedback on the communication
quality the participants are experiencing. Consider, for instance,
the simple scenario only involving two participants and a single RTP
session depicted in Figure 1:
+--------+ +---------+ +---------+
| |=== SSRC1 ===>| |=== SSRC3 ===>| |
| Alice | | B2BUA | | Bob |
| |<=== SSRC2 ===| |<=== SSRC4 ===| |
+--------+ +---------+ +---------+
Figure 1: B2BUA Modifying RTP Headers
In this common scenario, a participant (Alice) is communicating with
another participant (Bob) as a result of a signalling session managed
by a B2BUA: this B2BUA is also on the media path between the two and
is acting as a Media Relay. This means that two separate RTP
sessions are involved (one per side), each carrying two RTP streams
(one per media direction). As part of this process, the B2BUA is
also rewriting some of the RTP header information on the way. In
this example, just the Synchronization Source (SSRC) of the incoming
RTP streams is changed, but more information may be modified as well
(e.g., sequence numbers, timestamps, etc.). In particular, whenever
Alice sends an RTP packet, she sets her SSRC (SSRC1) in the RTP
header of her RTP source stream. The B2BUA rewrites the SSRC (SSRC3)
before relaying the packet to Bob. At the same time, RTP packets
sent by Bob (SSRC4) get their SSRC rewritten as well (SSRC2) before
being relayed to Alice.
Assuming now that Alice needs to inform Bob that she has lost several
packets in the last few seconds, she will place the related received
RTP stream SSRC she is aware of (SSRC2) together with her own (SSRC1)
in RTCP Reports and/or NACKs. Since the B2BUA is making use of
different SSRCs for the RTP streams in the RTP session it established
with each participant, blindly relaying Alice's incoming RTCP
messages to Bob would cause issues. These RTCP messages would
reference SSRCs Bob doesn't know about, which would result in
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precious feedback being dropped. In fact, Bob is only aware of SSRC4
(the one his source RTP stream uses) and SSRC3 (the one he's
receiving from the B2BUA in the received RTP stream) and knows
nothing about SSRC1 and SSRC2 in the messages he received instead.
Considering the feedback being dropped because of this may contain
precious information (e.g., related to packet loss, congestion, and
other network issues or considerations), the inability to take them
into account may lead to severe issues. For instance, Bob may flood
Alice with more media packets she can handle and/or not retransmit
the packets she missed and asked for. This may easily lead to a very
bad communication experience, if not eventually to an unwanted
termination of the communication itself.
This is just a trivial example that, together with additional
scenarios, will be addressed in the following sections.
Nevertheless, it is a valid example of how such a simple mishandling
of precious information may lead to serious consequences. This is
especially true if we picture more complex scenarios involving
several participants at the same time, multiple RTP sessions (e.g., a
video stream along audio) rather than a single one, redundancy RTP
streams, SSRC multiplexing, and so on. Considering how common B2BUA
deployments are, it is very important for them to properly address
RTCP messages in order to be sure that their activities on the media
plane do not break or interfere with anything relevant to the
session.
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].
In addition, this document uses, where relevant, the RTP-related
terminology defined in [RFC7656].
3. Signalling/Media Plane B2BUAs
As described in the Introduction (Section 1), it's very common for
B2BUA deployments to act on the media plane rather than just on the
signalling plane alone. In particular, [RFC7092] describes three
different categories of such B2BUAs: (1) a simple Media Relay that is
effectively unaware of anything that is transported; (2) a Media-
aware Relay that inspects and/or modifies RTP and RTCP messages as
they flow by; and (3) a full-fledged media termination entity that
terminates and generates RTP and RTCP messages as needed.
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[RFC3550] and [RFC7667] already mandate some specific behaviours in
the presence of certain topologies. However, due to their mixed
nature, B2BUAs sometimes can't or won't implement all relevant
specifications. This means that it's not rare to encounter issues
that may be avoided with more disciplined behaviour in that regard,
that is, if the B2BUAs followed at least a set of guidelines to
ensure no known problems occur. For this reason, the following
subsections describe the proper behaviour that B2BUAs, whatever above
category they fall in, should follow in order not to impact any end-
to-end RTCP effectiveness.
3.1. Media Relay
A Media Relay, as identified in [RFC7092], simply forwards all RTP
and RTCP messages it receives without either inspecting or modifying
them. Using the terminology in "RTP Topologies" [RFC7667], this can
be seen as an RTP Transport Translator. As such, B2BUAs acting as
Media Relays are not aware of what traffic they're handling. This
means that both packet payloads and packet headers are opaque to
them. Many Session Border Controllers (SBCs) implement this kind of
behaviour, e.g., when acting as a bridge between an inner and outer
network.
Considering that all headers and identifiers in both RTP and RTCP are
left untouched, issues like the SSRC mismatch described in the
previous section would not occur. However, similar problems could
still happen for different reasons, for instance, if the session
description prepared by the B2BUA, whether it has been modified or
not, ends up providing incorrect information. This may happen, for
example, if the Session Description Protocol (SDP) on either side
contains 'ssrc' [RFC5576] attributes that don't match the actual SSRC
being advertised on the media plane or if the B2BUA advertised
support for NACK because it implements it while the original INVITE
didn't. Such issues might occur, for instance, when the B2BUA acting
as a Media Relay is generating a new session description when
bridging an incoming call rather than using the original session
description. This may cause participants to find a mismatch between
the SSRCs advertised in the SDP and the ones actually observed in RTP
and RTCP messages or to have them either ignore or generate RTCP
feedback packets that were not explicitly advertised as supported.
In order to prevent such an issue, a Media Relay B2BUA SHOULD forward
all the SSRC- and RTCP-related SDP attributes when handling a
multimedia session setup between participants: this includes
attributes like 'ssrc' [RFC3261], 'rtcp-fb' [RFC4585], 'rtcp-xr-
attrib' [RFC3611], and others. However, certain SDP attributes may
lead to call failures when forwarded by a Media Relay, as they have
an implied assumption that the attribute describes the immediate
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peer. A clear example of this is the 'rtcp' [RFC3605] attribute,
which describes the expected RTCP peer port. Other attributes might
include the immediate peer's IP address, preferred transport, etc.
In general, the guideline is to require rewriting of attributes that
are implicitly describing the immediate peer. B2BUAs SHOULD forward
all other SDP attributes in order to avoid breaking additional
functionality that endpoints may be relying on. If implementors have
doubts about whether this guidance applies to a specific attribute,
they should test to determine if call failures occur.
The cited 'rtcp' example is also relevant whenever RTP/RTCP
multiplexing [RFC5761] support is being negotiated. If the B2BUA
acting as a Media Relay is unaware of the specifics of the traffic it
is handling, and as such may not have RTP/RTCP parsing capabilities,
it SHOULD reject RTP/RTCP multiplexing by removing the 'rtcp-mux' SDP
attribute. If instead the Media Relay is able to parse RTP/RTCP, and
can verify that demultiplexing can be performed without any RTP
Payload Type rewrites (i.e., no overlap between any RTP Payload Types
and the RTCP Payload Type space has been detected), then the B2BUA
SHOULD negotiate RTP/RTCP multiplexing support if advertised.
It is worth mentioning that, by leaving RTCP messages untouched, a
Media Relay may also leak information that, according to policies,
may need to be hidden or masqueraded, e.g., domain names in CNAME
items. Besides, these CNAME items may actually contain IP addresses:
this means that, should a NAT be involved in the communication, this
may actually result in CNAME collisions, which could indeed break the
end-to-end RTCP behaviour. While [RFC7022] can prevent this from
happening, there may be implementations that don't make use of it.
As such, a B2BUA MAY rewrite CNAME items if any potential collision
is detected, even in the Media Relay case. If a B2BUA does indeed
decide to rewrite CNAME items, then it MUST generate new CNAMEs
following [RFC7022]. The same SHOULD be done if RTP extensions
involving CNAMEs are involved (e.g., "urn:ietf:params:rtp-
hdrext:sdes:cname" [RFC7941]). If that is not possible, e.g.,
because the Media Relay does not have RTP header editing capabilities
or does not support these extensions, then the B2BUA MUST reject the
negotiation of such extensions when negotiating the session.
3.2. Media-Aware Relay
A Media-aware Relay, unlike the Media Relay addressed in the previous
section, is aware of the media traffic it is handling. This means it
inspects RTP and RTCP messages flowing by and may even modify their
headers. Using the terminology in [RFC3550], this can be seen as an
RTP Translator. A B2BUA implementing this role typically does not
inspect the RTP payloads, which would be opaque to them: this means
that the actual media would not be manipulated (e.g., transcoded).
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This makes them quite different from the Media Relays previously
discussed, especially in terms of the potential issues that may occur
at the RTCP level. In fact, being able to modify the RTP and RTCP
headers, such B2BUAs may end up modifying RTP-related information
like SSRC / Contributing Source (CSRC), sequence numbers, timestamps,
and others in an RTP stream before forwarding the modified packets to
the other interested participants. This means that, if not properly
disciplined, such behaviour may easily lead to issues like the one
described in the introductory section. For this reason, it is very
important for a B2BUA modifying RTP-related information across two
related RTP streams to also modify, in a coherent way, the same
information in RTCP messages.
It is worthwhile to point out that such a B2BUA may not necessarily
forward all the packets it receives. Selective Forwarding Units
(SFUs) [RFC7667], for instance, may be implemented to aggregate or
drop incoming RTCP messages while at the same time originating new
ones on their own. It is important to clarify that a B2BUA SHOULD
NOT randomly drop or forward RTCP feedback of the same type (e.g., a
specific XR block type or specific Feedback messages) within the
context of the same session as that may lead to confusing, if not
broken, feedback to the recipients of the message due to gaps in the
communication. As to the messages that are forwarded and/or
aggregated, it's important to make sure the information is coherent.
Besides the behaviour already mandated for RTCP translators in
Section 7.2 of [RFC3550], a media-aware B2BUA MUST handle incoming
RTCP messages to forward following these guidelines:
Sender Report (SR) [RFC3550]:
If the B2BUA has changed the SSRC of the sender RTP stream a
Sender Report refers to, it MUST update the SSRC in the SR packet
header as well. If the B2BUA has changed the SSRCs of other RTP
streams too, and any of these streams are addressed in any of the
SR Report Blocks, it MUST update the related values in the SR
Report Blocks as well. If the B2BUA has also changed the base RTP
sequence number when forwarding RTP packets, then this change MUST
be reflected in the 'extended highest sequence number received'
field in the Report Blocks. In case the B2BUA is acting as a
Selective Forwarding Unit (SFU) [RFC7667], it needs to track in
the outgoing SR, the relevant number of packets sent, and the
total amount of bytes sent to the receiver.
Receiver Report (RR) [RFC3550]:
The guidelines for SR apply to RR as well.
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Source Description (SDES) [RFC3550]:
If the B2BUA has changed the SSRC of any RTP stream addressed in
any of the chunks of an incoming SDES message, it MUST update the
related SSRCs in all the chunks. The same considerations made
with respect to CNAME collisions at the end of Section 3.1 apply
here as well.
BYE [RFC3550]:
If the B2BUA has changed the SSRC of any RTP stream addressed in
the SSRC/CSRC identifiers included in a BYE packet, it MUST update
them in the message.
APP [RFC3550]:
If the B2BUA has changed the SSRC of any RTP stream addressed in
the header of an APP packet, it MUST update the identifier in the
message. Should the B2BUA be aware of any specific APP message
format that contains additional information related to SSRCs, it
SHOULD update them accordingly as well.
Extended Reports (XRs) [RFC3611]:
If the B2BUA has changed the SSRC of the RTP stream associated
with the originator of an XR packet, it MUST update the SSRC in
the XR message header. The same guidelines given for SR/RR, with
respect to SSRC identifiers in Report Blocks, apply to all the
Report Block types in the XR message as well. If the B2BUA has
also changed the base RTP sequence number when forwarding RTP
packets, then this change MUST be reflected in the 'begin_seq' and
'end_seq' fields that are available in most of the Report Block
types that are part of the XR specification.
Receiver Summary Information (RSI) [RFC5760]:
If the B2BUA has changed any SSRC of RTP streams addressed in an
RSI packet, it MUST update the SSRC identifiers in the message.
This includes the distribution source SSRC, which MUST be
rewritten with the one the B2BUA uses to send RTP packets to each
sender participant, the summarized SSRC, and when a Collision Sub-
Report Block is available, the SSRCs in the related list.
Port Mapping (TOKEN) [RFC6284]:
If the B2BUA has changed any SSRC of RTP streams addressed in a
TOKEN packet, it MUST update the SSRC identifiers in the message.
This includes the Packet Sender SSRC, which MUST be rewritten with
the one the B2BUA uses to send RTP packets to each sender
participant, and the Requesting Client SSRC when the message is a
response, which MUST be rewritten using the related sender
participant(s) SSRC.
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Feedback Messages [RFC4585]:
All Feedback messages have a common packet format, which includes
the SSRC identifier of the Packet Sender and the SSRC identifier
of the media source the feedback is related to. Just as described
for the previous messages, these SSRC identifiers MUST be updated
in the message if the B2BUA has changed the SSRC of the RTP
streams addressed there. It MUST NOT, however, change a media
source SSRC that was originally set to zero, unless zero is
actually the SSRC that was chosen by one of the involved
endpoints, in which case the above-mentioned rules as to SSRC
rewriting apply. Considering that many Feedback messages also
include additional data as part of their specific Feedback Control
Information (FCI), a media-aware B2BUA MUST take care of them
accordingly, if it can parse and regenerate them, according to the
following guidelines:
NACK [RFC4585]:
A media-aware B2BUA MUST properly rewrite the Packet ID (PID)
of all addressed lost packets in the NACK FCI if it changed the
RTP sequence numbers.
TMMBR/TMMBN/FIR/TSTR/TSTN/VBCM [RFC5104]:
A media-aware B2BUA MUST properly rewrite the additional SSRC
identifier in the specific FCI if it changed the related RTP
SSRC of the media sender.
Receiver Estimated Max Bitrate (REMB) [RTCP-REMB]:
[RTCP-REMB] describes an RTCP payload-specific Feedback message
that reports the receiver's available bandwidth to the sender.
As of the time of this writing, REMB has been widely deployed
but has not been standardized. The REMB mechanism will not
function correctly across a media-aware B2BUA that changes the
SSRC of the media sender unless it also changes the SSRC values
in the REMB packet.
Explicit Congestion Notification (ECN) [RFC6679]:
The same guidelines given for SR/RR management apply,
considering the presence of sequence numbers in the ECN
Feedback Report format. For the management of RTCP XR ECN
Summary Report messages, the same guidelines given for generic
XR messages apply.
Apart from the generic guidelines related to Feedback messages, no
additional modifications are needed for Picture Loss Indication
(PLI), Slice Lost Indication (SLI), and Reference Picture Selection
Indication (RPSI) Feedback messages.
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Of course, the same considerations about the need for SDP and RTP/
RTCP information to be coherent applies to media-aware B2BUAs. This
means that, if a B2BUA changes any SSRC, it MUST update the related
'ssrc' attributes, if present, before sending it to the recipient.
Besides, it MUST rewrite the 'rtcp' attribute if provided. At the
same time, while a media-aware B2BUA is typically able to inspect/
modify RTCP messages, it may not support all RTCP messages. This
means that a B2BUA may choose to drop RTCP messages it can't parse.
In that case, a media-aware B2BUA MUST advertise its RTCP level of
support in the SDP in a coherent way in order to prevent, for
instance, a UAC from sending NACK messages that would never reach the
intended recipients. It's important to point out that, in case a
compound RTCP packet was received and any RTCP message in it needs to
be dropped, then the B2BUA SHOULD NOT drop the whole compound RTCP
packet, but only the selected messages.
The same considerations on CNAMEs made in regard to Media Relays
apply to Media-aware Relays as well. Specifically, if RTP extensions
involving CNAMEs are involved (e.g., "urn:ietf:params:rtp-
hdrext:sdes:cname" [RFC7941]) and negotiated because the B2BUA
supports them, then the B2BUA MUST update the CNAME value in there as
well, if it was changed. It is worth pointing out that, if the new
CNAME is larger than the old one, this would result in a larger RTP
packet than originally received. If the length of the updated packet
exceeds the MTU of any of the networks the packet will traverse, this
can result in the packet being dropped and lost by the recipient.
A different set of considerations is worthwhile for RTP/RTCP
multiplexing [RFC5761] and Reduced-Size RTCP [RFC5506]. While the
former allows for a better management of network resources by
multiplexing RTP packets and RTCP messages over the same transport,
the latter allows for a compression of RTCP messages, thus leading to
less network traffic. For RTP/RTCP multiplexing, a B2BUA acting as a
Media Relay may use it on either RTP session independently. This
means that, for instance, a Media Relay B2BUA may use RTP/RTCP
multiplexing on one side of the communication and not use it on the
other side, if the endpoint does not support it. This allows for a
better management of network resources on the side that does support
it. In case any of the parties in the communications supports it and
the B2BUA does too, the related 'rtcp-mux' SDP attribute MUST be
forwarded on the other side(s). If the B2BUA detects that any of the
parties in the communication do not support the feature, it may
decide to either disable it entirely or still advertise it for the
RTP sessions with parties that do support it. In case the B2BUA
decides to involve RTP/RTCP multiplexing, it MUST ensure that there
are no conflicting RTP Payload Type numbers on either side. When
there are, it MUST rewrite RTP Payload Type numbers to prevent
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conflicts in the session where the RTP/RTCP multiplexing is applied.
Should RTP Payload Types be rewritten, the related information in the
SDP MUST be updated accordingly.
For Reduced-Size RTCP, the considerations are a bit different. In
fact, while a Media Relay B2BUA may choose to use it on the side that
supports it and not on the side that doesn't, there are several
reasons for discouraging such behaviour. While Reduced-Size allows
for less network traffic related to RTCP messaging in general, this
gain may lead a Reduced-Size RTCP implementation to also issue a
higher rate of RTCP Feedback messages. This would result in
increased RTCP traffic on the side that does not support Reduced-Size
and could, as a consequence, actually be counterproductive if the
available bandwidth is different on the two sides. Negotiating a
session with both sides would allow the B2BUA to discover which one
supports Reduced-Size and which doesn't and decide whether or not to
allow the sides to independently use Reduced-Size. Should the B2BUA
decide to disable the feature on all sides, which is suggested in
case Reduced-Size is not supported by all parties involved, it MUST
NOT advertise support for the Reduced-Size RTCP functionality on
either side, by removing the 'rtcp-rsize' attribute from the SDP.
3.3. Media Terminator
A Media Terminator B2BUA, unlike simple Media Relays and media-aware
ones, is able to terminate media itself. As such, it can inspect
and/or modify RTP payloads as well. This means that such components,
for instance, can act as media transcoders and/or originate specific
RTP media. Using the terminology in "RTP Topologies" [RFC7667], this
can be seen as an RTP Media Translator. Such a topology can also be
seen as a back-to-back RTP session through a middlebox, as described
in Section 3.2.2 of [RFC7667]. Such a capability makes them quite
different from the previously introduced B2BUA typologies. Since
such a B2BUA would terminate RTP itself, it can take care of the
related statistics and feedback functionality directly, with no need
to simply relay any message between the participants in the
multimedia session.
For this reason, no specific guideline is needed to ensure a proper
end-to-end RTCP behaviour in such scenarios, because most of the
time, there would be no end-to-end RTCP interaction among the
involved participants in the first place. Nevertheless, should any
RTCP message actually need to be forwarded to another participant in
the multimedia session, the same guidelines provided for the media-
aware B2BUA case apply.
For RTP/RTCP multiplexing support, the same considerations already
given for the Media Relay management also apply to Media Terminators.
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Some different considerations might be given as to the Reduced-Size
RTCP functionality instead. In fact, in the Media Terminator case,
it is safe to use the feature independently on each side, as the
B2BUA would terminate RTCP. In that case, the B2BUA SHOULD advertise
and negotiate support for Reduced-Size if available and MUST NOT
otherwise.
4. IANA Considerations
This document does not require any IANA actions.
5. Security Considerations
The discussion in the previous sections on the management of RTCP
messages by a B2BUA worked under the assumption that the B2BUA has
actual access to the RTP/RTCP information itself. This is indeed
true if we assume that plain RTP and RTCP are being handled, but they
may not be once any security is enforced on RTP packets and RTCP
messages by means of Secure RTP (SRTP) [RFC3711].
While typically not an issue in the Media Relay case, where RTP and
RTCP packets are forwarded without any modification regardless of
whether or not security is involved, this could definitely have an
impact on Media-aware Relays and Media Terminator B2BUAs. As simple
example, if we envisage an SRTP / Secure RTCP (SRTCP) session across
a B2BUA where the B2BUA itself has no access to the keys used to
secure the session, there would be no way to manipulate SRTP headers
without violating the hashing on the packet. At the same time, there
would be no way to rewrite the RTCP information accordingly either.
For this reason, it is important to point out that the operations
described in the previous sections are only possible if the B2BUA has
a way to effectively manipulate the packets and messages flowing by.
This means that, when media security is involved, only the Media
Relay scenario can be properly addressed. Attempting to cover Media-
aware Relay and Media Termination scenarios when involving secure
sessions will inevitably lead to the B2BUA acting as a man in the
middle; consequently, its behaviour is unspecified and discouraged.
More considerations on this are provided in [RFC7879].
It is also worth pointing out that there are scenarios where an
improper management of RTCP messaging across a B2BUA may lead,
willingly or not, to situations not unlike an attack. As a simple
example, improper management of an REMB Feedback message containing,
e.g., information on the limited bandwidth availability for a user,
may lead to missing or misleading information to its peer. This may
cause the peer to increase the encoder bitrate, maybe up to a point
where a user with poor connectivity will inevitably be choked by an
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amount of data it cannot process. This scenario may thus result in
what looks like a Denial-of-Service (DoS) attack towards the user.
6. References
6.1. Normative References
[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>.
[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>.
[RFC3611] Friedman, T., Ed., Caceres, R., Ed., and A. Clark, Ed.,
"RTP Control Protocol Extended Reports (RTCP XR)",
RFC 3611, DOI 10.17487/RFC3611, November 2003,
<http://www.rfc-editor.org/info/rfc3611>.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
July 2006, <http://www.rfc-editor.org/info/rfc4566>.
[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>.
[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>.
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[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, DOI 10.17487/RFC5506, April
2009, <http://www.rfc-editor.org/info/rfc5506>.
[RFC5760] Ott, J., Chesterfield, J., and E. Schooler, "RTP Control
Protocol (RTCP) Extensions for Single-Source Multicast
Sessions with Unicast Feedback", RFC 5760,
DOI 10.17487/RFC5760, February 2010,
<http://www.rfc-editor.org/info/rfc5760>.
[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>.
[RFC6284] Begen, A., Wing, D., and T. Van Caenegem, "Port Mapping
between Unicast and Multicast RTP Sessions", RFC 6284,
DOI 10.17487/RFC6284, June 2011,
<http://www.rfc-editor.org/info/rfc6284>.
[RFC6679] Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P.,
and K. Carlberg, "Explicit Congestion Notification (ECN)
for RTP over UDP", RFC 6679, DOI 10.17487/RFC6679, August
2012, <http://www.rfc-editor.org/info/rfc6679>.
[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>.
[RFC7656] Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and
B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms
for Real-Time Transport Protocol (RTP) Sources", RFC 7656,
DOI 10.17487/RFC7656, November 2015,
<http://www.rfc-editor.org/info/rfc7656>.
[RFC7941] Westerlund, M., Burman, B., Even, R., and M. Zanaty, "RTP
Header Extension for the RTP Control Protocol (RTCP)
Source Description Items", RFC 7941, DOI 10.17487/RFC7941,
August 2016, <http://www.rfc-editor.org/info/rfc7941>.
6.2. Informative References
[RFC3605] Huitema, C., "Real Time Control Protocol (RTCP) attribute
in Session Description Protocol (SDP)", RFC 3605,
DOI 10.17487/RFC3605, October 2003,
<http://www.rfc-editor.org/info/rfc3605>.
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[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>.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009,
<http://www.rfc-editor.org/info/rfc5576>.
[RFC7092] Kaplan, H. and V. Pascual, "A Taxonomy of Session
Initiation Protocol (SIP) Back-to-Back User Agents",
RFC 7092, DOI 10.17487/RFC7092, December 2013,
<http://www.rfc-editor.org/info/rfc7092>.
[RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
DOI 10.17487/RFC7667, November 2015,
<http://www.rfc-editor.org/info/rfc7667>.
[RFC7879] Ravindranath, R., Reddy, T., Salgueiro, G., Pascual, V.,
and P. Ravindran, "DTLS-SRTP Handling in SIP Back-to-Back
User Agents", RFC 7879, DOI 10.17487/RFC7879, May 2016,
<http://www.rfc-editor.org/info/rfc7879>.
[RTCP-REMB]
Alvestrand, H., Ed., "RTCP message for Receiver Estimated
Maximum Bitrate", Work in Progress, draft-alvestrand-
rmcat-remb-03, October 2013.
Acknowledgements
The authors would like to thank Flavio Battimo and Pierluigi Palma
for their invaluable feedback in the early stages of this document.
The authors would also like to thank Colin Perkins, Bernard Aboba,
Albrecht Schwarz, Hadriel Kaplan, Keith Drage, Jonathan Lennox,
Stephen Farrell, Magnus Westerlund, Simon Perreault, and Ben Campbell
for their constructive comments, suggestions, and reviews that were
critical to the formulation and refinement of this document.
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Authors' Addresses
Lorenzo Miniero
Meetecho
Email: lorenzo@meetecho.com
Sergio Garcia Murillo
Medooze
Email: sergio.garcia.murillo@gmail.com
Victor Pascual
Oracle
Email: victor.pascual.avila@oracle.com
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