AVTCORE J. Lennox
Internet-Draft Vidyo
Updates: 3550, 4585 (if approved) M. Westerlund
Intended status: Standards Track Ericsson
Expires: January 04, 2015 Q. Wu
Huawei
C. Perkins
University of Glasgow
July 03, 2014
Sending Multiple Media Streams in a Single RTP Session
draft-ietf-avtcore-rtp-multi-stream-05
Abstract
This memo expands and clarifies the behaviour of Real-time Transport
Protocol (RTP) endpoints that use multiple synchronization sources
(SSRCs). This occurs, for example, when an endpoint sends multiple
media streams in a single RTP session. This memo updates RFC 3550
with regards to handling multiple SSRCs per endpoint in RTP sessions,
with a particular focus on RTCP behaviour. It also updates RFC 4585
to update and clarify the calculation of the timeout of SSRCs and the
inclusion of feedback messages.
Status of This Memo
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Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the
document authors. All rights reserved.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Use Cases For Multi-Stream Endpoints . . . . . . . . . . . . 3
3.1. Endpoints with Multiple Capture Devices . . . . . . . . . 3
3.2. Multiple Media Types in a Single RTP Session . . . . . . 3
3.3. Multiple Stream Mixers . . . . . . . . . . . . . . . . . 4
3.4. Multiple SSRCs for a Single Media Source . . . . . . . . 4
4. Use of RTP by endpoints that send multiple media streams . . 5
5. Use of RTCP by Endpoints that send multiple media streams . . 5
5.1. RTCP Reporting Requirement . . . . . . . . . . . . . . . 5
5.2. Initial Reporting Interval . . . . . . . . . . . . . . . 5
5.3. Aggregation of Reports into Compound RTCP Packets . . . . 6
5.3.1. Maintaining AVG_RTCP_SIZE . . . . . . . . . . . . . . 7
5.3.2. Scheduling RTCP with Multiple Reporting SSRCs . . . . 8
5.4. Use of RTP/AVPF Feedback . . . . . . . . . . . . . . . . 10
5.4.1. Choice of SSRC for Feedback Packets . . . . . . . . . 10
5.4.2. Scheduling an RTCP Feedback Packet . . . . . . . . . 11
6. RTCP Considerations for Streams with Disparate Rates . . . . 12
6.1. Timing out SSRCs . . . . . . . . . . . . . . . . . . . . 13
6.1.1. AVPF T_rr_interval Behaviour . . . . . . . . . . . . 13
6.1.2. Avoiding Premature Timeout . . . . . . . . . . . . . 14
6.1.3. RTP/AVP and RTP/AVPF Interoperability . . . . . . . . 15
6.1.4. Specified Behaviour . . . . . . . . . . . . . . . . . 16
6.2. Tuning RTCP transmissions . . . . . . . . . . . . . . . . 16
6.2.1. RTP/AVP and RTP/SAVP . . . . . . . . . . . . . . . . 16
6.2.2. RTP/AVPF and RTP/SAVPF . . . . . . . . . . . . . . . 18
7. Security Considerations . . . . . . . . . . . . . . . . . . . 19
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 19
9. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . . 19
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 20
10.1. Normative References . . . . . . . . . . . . . . . . . . 20
10.2. Informative References . . . . . . . . . . . . . . . . . 20
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 22
1. Introduction
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At the time the Real-Time Transport Protocol (RTP) [RFC3550] was
originally designed, and for quite some time after, endpoints in RTP
sessions typically only transmitted a single media stream, and thus
used a single synchronization source (SSRC) per RTP session, where
separate RTP sessions were typically used for each distinct media
type. Recently, however, a number of scenarios have emerged in which
endpoints wish to send multiple RTP media streams, distinguished by
distinct RTP synchronization source (SSRC) identifiers, in a single
RTP session. These are outlined in Section 3. Although the initial
design of RTP did consider such scenarios, the specification was not
consistently written with such use cases in mind. The specifications
are thus somewhat unclear.
This memo updates [RFC3550] to clarify behaviour in use cases where
endpoints use multiple SSRCs. It also updates [RFC4585] in regards
to the timeout of inactive SSRCs to resolve problematic behaviour as
well as clarifying the inclusion of feedback messages.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in RFC
2119 [RFC2119] and indicate requirement levels for compliant
implementations.
3. Use Cases For Multi-Stream Endpoints
This section discusses several use cases that have motivated the
development of endpoints that sends RTP data using multiple SSRCs in
a single RTP session.
3.1. Endpoints with Multiple Capture Devices
The most straightforward motivation for an endpoint to send multiple
simultaneous RTP streams in a session is the scenario where an
endpoint has multiple capture devices, and thus media sources, of the
same media type and characteristics. For example, telepresence
endpoints, of the type described by the CLUE Telepresence Framework
[I-D.ietf-clue-framework], often have multiple cameras or microphones
covering various areas of a room, and hence send several RTP streams.
3.2. Multiple Media Types in a Single RTP Session
Recent work has updated RTP
[I-D.ietf-avtcore-multi-media-rtp-session] and SDP
[I-D.ietf-mmusic-sdp-bundle-negotiation] to remove the historical
assumption in RTP that media sources of different media types would
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always be sent on different RTP sessions. In this work, a single
endpoint's audio and video RTP media streams (for example) are
instead sent in a single RTP session to reduce the number of
transport layer flows used.
3.3. Multiple Stream Mixers
There are several RTP topologies which can involve a central device
that itself generates multiple RTP media streams in a session. An
example is a mixer providing centralized compositing for a multi-
capture scenario like that described in Section 3.1. In this case,
the centralized node is behaving much like a multi-capturer endpoint,
generating several similar and related sources.
A more complex example is the selective forwarding middlebox,
described in Section 3.7 of [I-D.ietf-avtcore-rtp-topologies-update].
This is a middlebox that receives media streams from several
endpoints, and then selectively forwards modified versions of some
RTP streams toward the other endpoints to which it is connected. For
each connected endpoint, a separate media source appears in the
session for every other source connected to the middlebox,
"projected" from the original streams, but at any given time many of
them can appear to be inactive (and thus are receivers, not senders,
in RTP). This sort of device is closer to being an RTP mixer than an
RTP translator, in that it terminates RTCP reporting about the mixed
streams, and it can re-write SSRCs, timestamps, and sequence numbers,
as well as the contents of the RTP payloads, and can turn sources on
and off at will without appearing to be generating packet loss. Each
projected stream will typically preserve its original RTCP source
description (SDES) information.
3.4. Multiple SSRCs for a Single Media Source
There are also several cases where a single media source results in
the usage of multiple SSRCs within the same RTP session. Transport
robustness tools like RTP Retransmission [RFC4588] result in multiple
SSRCs, one with source data, and another with the repair data.
Scalable encoders and their RTP payload formats, like H.264's
extension for Scalable Video Coding(SVC) [RFC6190] can be transmitted
in a configuration where the scalable layers are distributed over
multiple SSRCs within the same session, to enable RTP packet stream
level (SSRC) selection and routing in conferencing middleboxes.
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4. Use of RTP by endpoints that send multiple media streams
Every RTP endpoint will have an allocated share of the available
session bandwidth, as determined by signalling and congestion
control. The endpoint MUST keep its total media sending rate within
this share. However, endpoints that send multiple media streams do
not necessarily need to subdivide their share of the available
bandwidth independently or uniformly to each media stream and its
SSRCs. In particular, an endpoint can vary the allocation to
different streams depending on their needs, and can dynamically
change the bandwidth allocated to different SSRCs (for example, by
using a variable rate codec), provided the total sending rate does
not exceed its allocated share. This includes enabling or disabling
media streams and their redundancy streams as more or less bandwidth
becomes available.
5. Use of RTCP by Endpoints that send multiple media streams
The RTP Control Protocol (RTCP) is defined in Section 6 of [RFC3550].
The description of the protocol is phrased in terms of the behaviour
of "participants" in an RTP session, under the assumption that each
endpoint is a participant with a single SSRC. However, for correct
operation in cases where endpoints can send multiple media streams,
the specification needs to be interpreted with each SSRC counting as
a participant in the session, so that an endpoint that has multiple
SSRCs counts as multiple participants. The following describes
several concrete cases where this applies.
5.1. RTCP Reporting Requirement
An RTP endpoint that has multiple SSRCs MUST treat each SSRC as a
separate participant in the RTP session, sending RTCP reports for
each of its SSRCs in every RTCP reporting interval. If the mechanism
in [I-D.ietf-avtcore-rtp-multi-stream-optimisation] is not used, then
each SSRC will send RTCP reports for all other SSRCs, including those
co-located at the same endpoint.
If the endpoint has some SSRCs that are sending data and some that
are only receivers, then they will receive different shares of the
RTCP bandwidth and calculate different base RTCP reporting intervals.
Otherwise, all SSRCs at an endpoint will calculate the same base RTCP
reporting interval. The actual reporting intervals for each SSRC are
randomised in the usual way, but reports can be aggregated as
described in Section 5.3.
5.2. Initial Reporting Interval
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When a participant joins a unicast session, the following text from
Section 6.2 of [RFC3550] applies: "For unicast sessions... the delay
before sending the initial compound RTCP packet MAY be zero." This
also applies to the individual SSRCs of an endpoint that has multiple
SSRCs, and such endpoints MAY send an initial RTCP packet for each of
their SSRCs immediately upon joining a unicast session.
Caution has to be exercised, however, when an endpoint (or middlebox)
with a large number of SSRCs joins a unicast session, since immediate
transmission of many RTCP reports can create a significant burst of
traffic, leading to transient congestion and packet loss due to queue
overflows. Implementers are advised to consider sending immediate
RTCP packets for only a small number of SSRCs (e.g., the one or two
SSRCs they consider most important), with the initial RTCP packets
for their other SSRCs being sent after the calculated initial RTCP
reporting interval, to avoid self congestion.
(tbd: is this recommendation sufficiently strong?)
5.3. Aggregation of Reports into Compound RTCP Packets
As outlined in Section 5.1, an endpoint with multiple SSRCs has to
treat each SSRC as a separate participant when it comes to sending
RTCP reports. This will lead to each SSRC sending a compound RTCP
packet in each reporting interval. Since these packets are coming
from the same endpoint, it might reasonably be expected that they can
be aggregated to reduce overheads. Indeed, Section 6.1 of [RFC3550]
allows RTP translators and mixers to aggregate packets in similar
circumstances:
"It is RECOMMENDED that translators and mixers combine individual
RTCP packets from the multiple sources they are forwarding into
one compound packet whenever feasible in order to amortize the
packet overhead (see Section 7). An example RTCP compound packet
as might be produced by a mixer is shown in Fig. 1. If the
overall length of a compound packet would exceed the MTU of the
network path, it SHOULD be segmented into multiple shorter
compound packets to be transmitted in separate packets of the
underlying protocol. This does not impair the RTCP bandwidth
estimation because each compound packet represents at least one
distinct participant. Note that each of the compound packets MUST
begin with an SR or RR packet."
The allows RTP translators and mixers to generate compound RTCP
packets that contain multiple SR or RR packets from different SSRCs,
as well as any of the other packet types. There are no restrictions
on the order in which the RTCP packets can occur within the compound
packet, except the regular rule that the compound RTCP packet starts
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with an SR or RR packet. Due to this rule, correctly implemented RTP
endpoints will be able to handle compound RTCP packets that contain
RTCP packets relating to multiple SSRCs.
Accordingly, endpoints that use multiple SSRCs MAY aggregate the RTCP
packets sent by their different SSRCs into compound RTCP packets,
provided they maintain the average RTCP packet size as described in
Section 5.3.1, and schedule packet transmission and aggregation as
described in Section 5.3.2.
5.3.1. Maintaining AVG_RTCP_SIZE
The RTCP scheduling algorithm in [RFC3550] works on a per-SSRC basis.
Each SSRC sends a single compound RTCP packet in each RTCP reporting
interval. When an endpoint uses multiple SSRCs, it is desirable to
aggregate the compound RTCP packets sent by its SSRCs, reducing the
overhead by forming a larger compound RTCP packet. This aggregation
can be done as described in Section 5.3.2, provided the average RTCP
packet size calculation is updated as follows.
Participants in an RTP session update their estimate of the average
RTCP packet size (avg_rtcp_size) each time they send or receive an
RTCP packet (see Section 6.3.3 of [RFC3550]). When a compound RTCP
packet that contains RTCP packets from several SSRCs is sent or
received, the avg_rtcp_size estimate for each SSRC that is reported
upon is updated using div_packet_size rather than the actual packet
size:
avg_rtcp_size = (1/16) * div_packet_size + (15/16) * avg_rtcp_size
where div_packet_size is packet_size divided by the number of SSRCs
reporting in that compound packet. The number of SSRCs reporting in
a compound packet is determined by counting the number of different
SSRCs that are the source of Sender Report (SR) or Receiver Report
(RR) RTCP packets within the compound RTCP packet. Non-compound RTCP
packets (i.e., RTCP packets that do not contain an SR or RR packet
[RFC5506]) are considered report on a single SSRC.
An SSRC doesn't follow the above rule, and instead uses the full RTCP
compound packet size to calculate avg_rtcp_size, will derive an RTCP
reporting interval that is overly large by a factor that is
proportional to the number of SSRCs aggregated into compound RTCP
packets and the size of set of SSRCs being aggregated relative to the
total number of participants. This increased RTCP reporting interval
can cause premature timeouts if it is more than five times the
interval chosen by the SSRCs that understand compound RTCP that
aggregate reports from many SSRCs. A 1500 octet MTU can fit six
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typical size reports into a compound RTCP packet, so this is a real
concern if endpoints aggregate RTCP reports from multiple SSRCs. If
compatibility with non-updated endpoints is a concern, the number of
reports from different SSRCs aggregated into a single compound RTCP
packet SHOULD be limited.
5.3.2. Scheduling RTCP with Multiple Reporting SSRCs
When implementing RTCP packet scheduling for cases where multiple
reporting SSRCs are aggregating their RTCP packets in the same
compound packet there are a number of challenges. First of all, we
have the goal of not changing the general properties of the RTCP
packet transmissions, which include the general inter-packet
distribution, and the behaviour for dealing with flash joins as well
as other dynamic events.
The below specified mechanism deals with:
o That one can't have a-priori knowledge about which RTCP packets
are to be sent, or their size, prior to generating the packets.
In which case, the time from generation to transmission ought to
be as short as possible to minimize the information that becomes
stale.
o That one has an MTU limit, that one ought to avoid exceeding, as
that requires lower-layer fragmentation (e.g., IP fragmentation)
which impacts the packets' probability of reaching the
receiver(s).
Schedule all the endpoint's local SSRCs individually for transmission
using the regular calculation of Tn for the profile being used. Each
time a SSRC's Tn timer expires, do the regular reconsideration. If
the reconsideration indicates that an RTCP packet is to be sent:
1. Consider if an additional SSRC can be added. That consideration
is done by picking the SSRC which has the Tn value closest in
time to now (Tc).
2. Calculate how much space for RTCP packets would be needed to add
that SSRC.
3. If the considered SSRC's RTCP Packets fit within the lower layer
datagram's Maximum Transmission Unit, taking the necessary
protocol headers into account and the consumed space by prior
SSRCs, then add that SSRC's RTCP packets to the compound packet
and go again to Step 1.
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4. If the considered SSRC's RTCP Packets will not fit within the
compound packet, then transmit the generated compound packet.
5. Update the RTCP Parameters for each SSRC that has been included
in the sent RTCP packet. The Tp value for each SSRC MUST be
updated as follows:
For the first SSRC: As this SSRC was the one that was
reconsidered the tp value is set to the tc as defined in
RTP [RFC3550].
For any additional SSRC: The tp value SHALL be set to the
transmission time this SSRC would have had it not been
aggregated and given the current existing session context.
This value is derived by taking this SSRC's Tn value and
performing reconsideration and updating tn until tp + T <=
tn. Then set tp to this tn value.
6. For the sent SSRCs calculate new tn values based on the updated
parameters and reschedule the timers.
Reverse reconsideration needs to be performed as specified in RTP
[RFC3550]. It is important to note that under the above algorithm
when performing reconsideration, the value of tp can actually be
larger than tc. However, that still has the desired effect of
proportionally pulling the tp value towards tc (as well as tn) as the
group size shrinks in direct proportion the reduced group size.
The above algorithm has been shown in simulations to maintain the
inter-RTCP-packet transmission distribution for the SSRCs and consume
the same amount of bandwidth as non-aggregated packets in RTP
sessions with static sets of participants. With this algorithm the
actual transmission interval for any SSRC triggering an RTCP compound
packet transmission is following the regular transmission rules. It
also handles the cases where the number of SSRCs that can be included
in an aggregated packet varies. An SSRC that previously was
aggregated and fails to fit in a packet still has its own
transmission scheduled according to normal rules. Thus, it will
trigger a transmission in due time, or the SSRC will be included in
another aggregate.
The algorithm's behaviour under SSRC group size changes is under
investigation. However, it is expected to be well behaved based on
the following analyses.
RTP sessions where the number of SSRC are growing: When the group
size is growing, the Td values grow in proportion to the number of
new SSRCs in the group. The reconsideration when the timer for
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the tn expires, that SSRC will reconsider the transmission and
with a certain probability reschedule the tn timer. This part of
the reconsideration algorithm is only impacted by the above
algorithm by having tp values that are in the future instead of
set to the time of the actual last transmission at the time of
updating tp. Thus the scheduling causes in worst case a plateau
effect for that SSRC. That effect depends on how far into the
future tp can advance.
RTP sessions where the number of SSRC are shrinking: When the group
shrinks, reverse reconsideration moves the tp and tn values
towards tc proportionally to the number of SSRCs that leave the
session compared to the total number of participants when they
left. Thus the also group size reductions need to be handled.
In general the potential issue that might exist depends on how far
into the future the tp value can drift compared to the actual packet
transmissions that occur. That drift can only occur for an SSRC that
never is the trigger for RTCP packet transmission and always gets
aggregated and where the calculated packet transmission interval
randomly occurs so that tn - tp for this SSRC is on average larger
than the ones that gets transmitted.
5.4. Use of RTP/AVPF Feedback
This section discusses the transmission of RTP/AVPF feedback packets
when the transmitting endpoint has multiple SSRCs.
5.4.1. Choice of SSRC for Feedback Packets
When an RTP/AVPF endpoint has multiple SSRCs, it can choose what SSRC
to use as the source for the RTCP feedback packets it sends. Several
factors can affect that choice:
o RTCP feedback packets relating to a particular media type SHOULD
be sent by an SSRC that receives that media type. For example,
when audio and video are multiplexed onto a single RTP session,
endpoints will use their audio SSRC to send feedback on the audio
received from other participants.
o RTCP feedback packets and RTCP codec control messages that are
notifications or indications regarding RTP data processed by an
endpoint MUST be sent from the SSRC used by that RTP data. This
includes notifications that relate to a previously received
request or command.
o If separate SSRCs are used to send and receive media, then the
corresponding SSRC SHOULD be used for feedback, since they have
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differing RTCP bandwidth fractions. This can also effect the
consideration if the SSRC can be used in immediate mode or not.
o Some RTCP feedback packet types requires consistency in the SSRC
used. For example, if one sets a TMMBR limitation, the same SSRC
needs to be used to remove the limitation.
When an RTCP feedback packet is sent as part of a compound RTCP
packet that aggregates reports from multiple SSRCs, there is no
requirement that the compound packet contains an SR or RR packet
generated by the sender of the RTCP feedback packet. For reduced-
size RTCP packets, aggregation of RTCP feedback packets from multiple
sources is not limited further than Section 4.2.2 of [RFC5506].
5.4.2. Scheduling an RTCP Feedback Packet
When an SSRC has a need to transmit a feedback packet in early mode
it follows the scheduling rules defined in Section 3.5 in RTP/AVPF
[RFC4585]. When following these rules the following clarifications
need to be taken into account:
o That a session is considered to be point-to-point or multiparty
not based on the number of SSRCs, but the number of endpoints
directly seen in the RTP session by the endpoint. tbd: Clarify
what is considered to "see" an endpoint?
o Note that when checking if there is already a scheduled compound
RTCP packet containing feedback messages (Step 2 in
Section 3.5.2), that check is done considering all local SSRCs.
TBD: The above does not allow an SSRC that is unable to send either
an early or regular RTCP packet with the feedback message within the
T_max_fb_delay to trigger another SSRC to send an early packet to
which it could piggyback. Nor does it allow feedback to piggyback on
even regular RTCP packet transmissions that occur within
T_max_fb_delay. A question is if either of these behaviours ought to
be allowed. The latter appears simple and straight forward. Instead
of discarding a FB message in step 4a: alternative 2, one could place
such messages in a cache with a discard time equal to T_max_fb_delay,
and in case any of the SSRCs schedule an RTCP packet for transmission
within that time, it includes this message. The former case can have
more widespread impact on the application, and possibly also on the
RTCP bandwidth consumption as it allows for more massive bursts of
RTCP packets. Still, on a time scale of a regular reporting
interval, it ought to have no effect on the RTCP bandwidth as the
extra feedback messages increase the avg_rtcp_size.
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6. RTCP Considerations for Streams with Disparate Rates
It is possible for a single RTP session to carry streams of greatly
differing bandwidth. There are two scenarios where this can occur.
The first is when a single RTP session carries multiple flows of the
same media type, but with very different quality; for example a video
switching multi-point conference unit might send a full rate high-
definition video stream of the active speaker but only thumbnails for
the other participants, all sent in a single RTP session. The second
scenarios occurs when audio and video flows are sent in a single RTP
session, as discussed in [I-D.ietf-avtcore-multi-media-rtp-session].
An RTP session has a single set of parameters that configure the
session bandwidth, the RTCP sender and receiver fractions (e.g., via
the SDP "b=RR:" and "b=RS:" lines), and the parameters of the RTP/
AVPF profile [RFC4585] (e.g., trr-int) if that profile (or its secure
extension, RTP/SAVPF [RFC5124]) is used. As a consequence, the RTCP
reporting interval will be the same for every SSRC in an RTP session.
This uniform RTCP reporting interval can result in RTCP reports being
sent more often than is considered desirable for a particular media
type. For example, if an audio flow is multiplexed with a high
quality video flow where the session bandwidth is configured to match
the video bandwidth, this can result in the RTCP packets having a
greater bandwidth allocation than the audio data rate. If the
reduced minimum RTCP interval described in Section 6.2 of [RFC3550]
is used in the session, which might be appropriate for video where
rapid feedback is wanted, the audio sources could be expected to send
RTCP packets more often than they send audio data packets. This is
most likely undesirable, and while the mismatch can be reduced
through careful tuning of the RTCP parameters, particularly trr_int
in RTP/AVPF sessions, it is inherent in the design of the RTCP timing
rules, and affects all RTP sessions containing flows with mismatched
bandwidth.
Having multiple media types in one RTP session also results in more
SSRCs being present in this RTP session. This increasing the amount
of cross reporting between the SSRCs. From an RTCP perspective, two
RTP sessions with half the number of SSRCs in each will be slightly
more efficient. If someone needs either the higher efficiency due to
the lesser number of SSRCs or the fact that one can't tailor RTCP
usage per media type, they need to use independent RTP sessions.
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When it comes to configuring RTCP the need for regular periodic
reporting needs to be weighted against any feedback or control
messages being sent. Applications using RTP/AVPF or RTP/SAVPF are
RECOMMENDED to consider setting the trr-int parameter to a value
suitable for the application's needs, thus potentially reducing the
need for regular reporting and thus releasing more bandwidth for use
for feedback or control.
Another aspect of an RTP session with multiple media types is that
the RTCP packets, RTCP Feedback Messages, or RTCP XR metrics used
might not be applicable to all media types. Instead, all RTP/RTCP
endpoints need to correlate the media type of the SSRC being
referenced in a message or packet and only use those that apply to
that particular SSRC and its media type. Signalling solutions might
have shortcomings when it comes to indicating that a particular set
of RTCP reports or feedback messages only apply to a particular media
type within an RTP session.
6.1. Timing out SSRCs
This section discusses issues around timing out SSRCs. After the
discussion, clarified and mandated behaviour for SSRC timeout is
specified.
6.1.1. AVPF T_rr_interval Behaviour
The RTP/AVPF profile includes a mechanism for suppressing regular
RTCP reporting from being sent unnecessarily frequently if sufficient
RTCP bandwidth is configured. This mechanism is defined in
Section 3.5.3 of [RFC4585], and can be summarized as follows: if less
than a randomized T_rr_interval value has passed since the last
regular report, and no feedback messages need to be sent, then the
RTCP regular report is suppressed. The randomization is done
linearly in the interval 0.5 to 1.5 times T_rr_interval. The
randomized T_rr_interval is recalculated after every transmitted
regular packet, i.e when t_rr_last was updated. The benefit of the
suppression mechanism is that it avoids wasting bandwidth when there
is nothing requiring frequent RTCP transmissions, but still allows
utilization of the configured bandwidth when feedback is needed.
Unfortunately this suppression mechanism has some behaviour that is
less than ideal. First of all, the randomized T_rr_interval is
distributed over a larger range than the actual transmission interval
for RTCP would be if T_rr_interval and Td had the same value. The
reconsideration mechanism and its compensation factor result in the
actual RTCP transmission intervals for a Td having a distribution
that is exponentially growing more likely with higher values, and is
bounded to the interval [0.5/1.21828, 1.5/1.21828]*Td, i.e. with a
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Td value of 5 s [2.052, 6.156]. In comparison, the suppression acts
in an interval that is 0.5 to 1.5 times the T_rr_interval, i.e. for
T_rr_interval = 5 s this is [2.5, 7.5].
The effect of the above is that the time period between two RTCP
packets when using T_rr_interval suppression can become very long
compared to the average input values. The longest time interval
between one transmitted regular RTCP compound packet and the next
when T_rr_interval suppression is being used are: First the maximum
T_rr_interval, i.e. 1.5*T_rr_interval. Assuming that the last
suppressed packet would have been sent at 1.5*T_rr_interval, the
maximum interval until a packet can be sent under the regular
scheduling is 1.5/1.21828*Td. Thus, the maximum time in total is
1.5*T_rr_interval + 1.5/1.21828*Td.
If Td and T_rr_interval have the same value, i.e. the minimal
interval desired (T_rr_interval) and the actual actual average
interval specified by the RTCP scheduling algorithm (Td) are the
same, one might expect that RTCP packets would be sent according to
the regular mechanism. Instead, this algorithm results in the RTCP
packets being sent anywhere from 0.5*Td to ~2.731*Td. The
probability distribution over that time is also non-trivial in its
shape, somewhat similar to a saw tooth.
Thus, we recommend that the AVPF regular transmission mechanism is
revised in the future. This issue also has further implications as
discussed in the next section.
6.1.2. Avoiding Premature Timeout
In RTP/AVP [RFC3550] the timeout behaviour is simple and is 5 times
Td, where Td is calculated with a Tmin value of 5 seconds. In other
words, if the RTCP bandwidth allowed for an RTCP interval more
frequent than every 5 seconds on average, then timeout happened after
5*Td = 25 seconds of no activity from the SSRC (RTP or RTCP),
otherwise it was 5 average reporting intervals.
RTP/AVPF [RFC4585] introduced two different behaviours depending on
the value of T_rr_interval. When T_rr_interval was 0, it defaulted
to the same Td calculation in RTP/AVP [RFC3550]. However, when
T_rr_interval is non-zero the Tmin value become T_rr_interval in that
calculation, most likely to enable speed up the detection of timed
out SSRCs. However, using a non-zero T_rr_interval has two
consequences for RTP behaviour.
First, the number of actually sent RTCP packets for an SSRC that
currently is not an active RTP sender can become very low due to the
issue discussed above in Section 6.1.1. As the RTCP packet interval
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can be as long as 2.73*Td, then during a 5*Td time period an endpoint
may in fact transmit only a single RTCP packet. The long intervals
result in fewer RTCP packets, to a point where a one or two packet
losses in RTCP result in timing out an SSRC.
Second, the change also increased RTP/AVPF's brittleness to both
packet loss and configuration errors. In many cases, when one
desires to use RTP/AVPF for its feedback, one will ensure that RTCP
is configured for more frequent transmissions on average than every 5
seconds. Thus, many more RTP and RTCP packets can be transmitted
during the time interval. Lets consider an implementation that would
follow the RTP/AVP or RTP/AVPF with T_rr_interval = 0 rules for
timeout, also when T_rr_interval is not zero. In such a case when
the configured value of the T_rr_interval is significantly smaller
than 5 seconds, e.g. less than 1 second, then a difference between
using 0.1 seconds and 0.6 seconds has no significant impact on when
an SSRC will be timed out. However, such a configuration difference
between two endpoints following RFC 4585 will result in that the
endpoint configured with T_rr_interval = 0.1 will frequently timeout
SSRCs currently not sending RTP, from the endpoint configured with
0.6, as that is six times the Td value used by the endpoint
configured with T_rr_interval=0.1, assuming sufficient bandwidth.
For this reason such a change is implemented below in Section 6.1.4.
6.1.3. RTP/AVP and RTP/AVPF Interoperability
If endpoints implementing the RTP/AVP and RTP/AVPF profiles (or their
secure variants) are combined in a single RTP session, and the RTP/
AVPF endpoints use a non-zero T_rr_interval that is significantly
lower than 5 seconds, then there is a risk that the RTP/AVPF
endpoints will prematurely timeout the RTP/AVP SSRCs due to their
different RTCP timeout intervals. Conversely, if the RTP/AVPF
endpoints use a T_rr_interval that is significant larger than 5
seconds, there is a risk that the RTP/AVP endpoints will timeout the
RTP/AVPF SSRCs.
If such mixed RTP profiles are used, (though this is NOT
RECOMMENDED), and the AVPF endpoint is not updated to follow this
specification, then the RTP/AVPF session SHOULD use a non-zero
T_rr_interval that is 4 seconds.
It might appear strange to use a T_rr_interval of 4 seconds. It
might be intuitive that this value ought to be 5 seconds, as then
both the RTP/AVP and RTP/AVPF would use the same timeout period.
However, considering regular RTCP transmission and their packet
intervals for RTP/AVPF its mean value will (with non-zero
T_rr_interval) be larger than T_rr_interval due to the scheduling
algorithm's behaviour as discussed in Section 6.1.1. Thus, to enable
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an equal amount of regular RTCP transmissions in each directions
between RTP/AVP and RTP/AVPF endpoints, taking the altered timeout
intervals into account, the optimal value is around four (4), where
almost four transmissions will on average occur in each direction
between the different profile types given an otherwise good
configuration of parameters in regards to T_rr_interval. If the RTCP
bandwidth parameters are selected so that Td based on bandwidth is
close to 4, i.e. close to T_rr_interval the risk increases that RTP/
AVPF SSRCs will be timed out by RTP/AVP endpoints, as the RTP/AVPF
SSRC might only manage two transmissions in the timeout period.
6.1.4. Specified Behaviour
The above considerations result in the following clarification and
RTP/AVPF specification change.
All SSRCs used in an RTP session MUST use the same timeout behaviour
to avoid premature timeouts. This will depend on the RTP profile and
its configuration. The RTP specification provides several options
that can influence the values used when calculating the time
interval. To avoid interoperability issues when using this
specification, this document makes several clarifications to the
calculations.
For RTP/AVP, RTP/SAVP, RTP/AVPF, and RTP/SAVPF, the timeout interval
SHALL be calculated using a multiplier of 5, i.e. the timeout
interval becomes 5*Td. The Td calculation SHALL be done using a Tmin
value of 5 seconds, not the reduced minimal interval even if used to
calculate RTCP packet transmission intervals. This changes the
behaviour for the RTP/AVPF or RTP/SAVPF profiles when T_rr_interval
!= 0, a behaviour defined in Section 3.5.4 of RFC 4585, i.e. Tmin in
the Td calculation is the T_rr_interval.
6.2. Tuning RTCP transmissions
This sub-section discusses what tuning can be done to reduce the
downsides of the shared RTCP packet intervals. First, it is
considered what possibilities exist for the RTP/AVP [RFC3551]
profile, then what additional tools are provided by RTP/AVPF
[RFC4585].
6.2.1. RTP/AVP and RTP/SAVP
When using the RTP/AVP or RTP/SAVP profiles the tuning one can do is
very limited. The controls one has are limited to the RTCP bandwidth
values and whether the minimum RTCP interval is scaled according to
the bandwidth. As the scheduling algorithm includes both random
factors and reconsideration, one can't simply calculate the expected
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average transmission interval using the formula for Td. But it does
indicate the important factors affecting the transmission interval,
namely the RTCP bandwidth available for the role (Active Sender or
Participant), the average RTCP packet size, and the number of SSRCs
classified in the relevant role. Note that if the ratio of senders
to total number of session participants is larger than the ratio of
RTCP bandwidth for senders in relation to the total RTCP bandwidth,
then senders and receivers are treated together.
Let's start with some basic observations:
a. Unless the scaled minimum RTCP interval is used, then Td prior to
randomization and reconsideration can never be less than 5
seconds (assuming default Tmin of 5 seconds).
b. If the scaled minimum RTCP interval is used, Td can become as low
as 360 divided by RTP Session bandwidth in kilobits. In SDP the
RTP session bandwidth is signalled using b=AS. An RTP Session
bandwidth of 72 kbps results in Tmin being 5 seconds. An RTP
session bandwidth of 360 kbps of course gives a Tmin of 1 second,
and to achieve a Tmin equal to once every frame for a 25 Hz video
stream requires an RTP session bandwidth of 9 Mbps! (The use of
the RTP/AVPF or RTP/SAVPF profile allows a smaller Tmin, and
hence more frequent RTCP reports, as discussed below).
c. Let's calculate the number (n) of SSRCs in the RTP session that
5% of the session bandwidth can support to yield a Td value equal
to Tmin with minimal scaling. For this calculation we have to
make two assumptions. The first is that we will consider most or
all SSRC being senders, resulting in everyone sharing the
available bandwidth. Secondly we will select an average RTCP
packet size. This packet will consist of an SR, containing (n-1)
report blocks up to 31 report blocks, and an SDES item with at
least a CNAME (17 bytes in size) in it. Such a basic packet will
be 800 bytes for n>=32. With these parameters, and as the
bandwidth goes up the time interval is proportionally decreased
(due to minimal scaling), thus all the example bandwidths 72
kbps, 360 kbps and 9 Mbps all support 9 SSRCs.
d. The actual transmission interval for a Td value is [0.5*Td/
1.21828,1.5*Td/1.21828], which means that for Td = 5 seconds, the
interval is actually [2.052,6.156] and the distribution is not
uniform, but rather exponentially-increasing. The probability
for sending at time X, given it is within the interval, is
probability of picking X in the interval times the probability to
randomly picking a number that is <=X within the interval with an
uniform probability distribution. This results in that the
majority of the probability mass is above the Td value.
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To conclude, with RTP/AVP and RTP/SAVP the key limitation for small
unicast sessions is going to be the Tmin value. Thus the RTP session
bandwidth configured in RTCP has to be sufficiently high to reach the
reporting goals the application has following the rules for the
scaled minimal RTCP interval.
6.2.2. RTP/AVPF and RTP/SAVPF
When using RTP/AVPF or RTP/SAVPF we get a quite powerful additional
tool, the setting of the T_rr_interval which has several effects on
the RTCP reporting. First of all as Tmin is set to 0 after the
initial transmission, the regular reporting interval is instead
determined by the regular bandwidth based calculation and the
T_rr_interval. This has the effect that we are no longer restricted
by the minimal interval or even the scaling rule for the minimal
rule. Instead the RTCP bandwidth and the T_rr_interval are the
governing factors.
Now it also becomes important to separate between the application's
need for regular reports and RTCP feedback packet types. In both
regular RTCP mode, as in Early RTCP Mode, the usage of the
T_rr_interval prevents regular RTCP packets, i.e. packets without
any Feedback packets, to be sent more often than T_rr_interval. This
value is applied to prevent any regular RTCP packet to be sent less
than T_rr_interval times a uniformly distributed random value from
the interval [0.5,1.5] after the previous regular packet packet. The
random value recalculated after each regular RTCP packet
transmission.
So applications that have a use for feedback packets for some media
streams, for example video streams, but don't want frequent regular
reporting for audio, could configure the T_rr_interval to a value so
that the regular reporting for both audio and video is at a level
that is considered acceptable for the audio. They could then use
feedback packets, which will include RTCP SR/RR packets, unless
reduced-size RTCP feedback packets [RFC5506] are used, and can
include other report information in addition to the feedback packet
that needs to be sent. That way the available RTCP bandwidth can be
focused for the use which provides the most utility for the
application.
Using T_rr_interval still requires one to determine suitable values
for the RTCP bandwidth value, in fact it might make it even more
important, as this is more likely to affect the RTCP behaviour and
performance than when using RTP/AVP, as there are fewer limitations
affecting the RTCP transmission.
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When using T_rr_interval, i.e. having it be non zero, there are
configurations that have to be avoided. If the resulting Td value is
smaller but close to T_rr_interval then the interval in which the
actual regular RTCP packet transmission falls into becomes very
large, from 0.5 times T_rr_interval up to 2.73 times the
T_rr_interval. Therefore for configuration where one intends to have
Td smaller than T_rr_interval, then Td is RECOMMENDED to be targeted
at values less than 1/4th of T_rr_interval which results in that the
range becomes [0.5*T_rr_interval, 1.81*T_rr_interval].
With RTP/AVPF, using a T_rr_interval of 0 or with another low value
significantly lower than Td still has utility, and different
behaviour compared to RTP/AVP. This avoids the Tmin limitations of
RTP/AVP, thus allowing more frequent regular RTCP reporting. In fact
this will result that the RTCP traffic becomes as high as the
configured values.
(tbd: a future version of this memo will include examples of how to
choose RTCP parameters for common scenarios)
There exists no method within the specification for using different
regular RTCP reporting intervals depending on the media type or
individual media stream.
7. Security Considerations
In the secure RTP protocol (SRTP) [RFC3711], the cryptographic
context of a compound SRTCP packet is the SSRC of the sender of the
first RTCP (sub-)packet. This could matter in some cases, especially
for keying mechanisms such as Mikey [RFC3830] which allow use of per-
SSRC keying.
Other than that, the standard security considerations of RTP apply;
sending multiple media streams from a single endpoint does not appear
to have different security consequences than sending the same number
of streams.
8. IANA Considerations
No IANA actions needed.
9. Open Issues
At this stage this document contains a number of open issues. The
below list tries to summarize the issues:
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1. Do we need to provide a recommendation for unicast session
joiners with many sources to not use 0 initial minimal interval
from bit-rate burst perspective?
2. RTCP parameters for common scenarios in Section 6.2?
3. Is scheduling algorithm working well with dynamic changes?
4. Are the scheduling algorithm changes impacting previous
implementations in such a way that the report aggregation has to
be agreed on, and thus needs to be considered as an optimization?
5. An open question is if any improvements or clarifications ought
to be allowed regarding FB message scheduling in multi-SSRC
endpoints.
10. References
10.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July
2006.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, February 2008.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009.
10.2. Informative References
[I-D.ietf-avtcore-multi-media-rtp-session]
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Westerlund, M., Perkins, C., and J. Lennox, "Sending
Multiple Types of Media in a Single RTP Session", draft-
ietf-avtcore-multi-media-rtp-session-05 (work in
progress), February 2014.
[I-D.ietf-avtcore-rtp-multi-stream-optimisation]
Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
"Sending Multiple Media Streams in a Single RTP Session:
Grouping RTCP Reception Statistics and Other Feedback ",
draft-ietf-avtcore-rtp-multi-stream-optimisation-00 (work
in progress), July 2013.
[I-D.ietf-avtcore-rtp-topologies-update]
Westerlund, M. and S. Wenger, "RTP Topologies", draft-
ietf-avtcore-rtp-topologies-update-02 (work in progress),
May 2014.
[I-D.ietf-clue-framework]
Duckworth, M., Pepperell, A., and S. Wenger, "Framework
for Telepresence Multi-Streams", draft-ietf-clue-
framework-16 (work in progress), June 2014.
[I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
negotiation-07 (work in progress), April 2014.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
[RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control
Protocol Extended Reports (RTCP XR)", RFC 3611, November
2003.
[RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
August 2004.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
July 2006.
[RFC6190] Wenger, S., Wang, Y.-K., Schierl, T., and A.
Eleftheriadis, "RTP Payload Format for Scalable Video
Coding", RFC 6190, May 2011.
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Authors' Addresses
Jonathan Lennox
Vidyo, Inc.
433 Hackensack Avenue
Seventh Floor
Hackensack, NJ 07601
USA
Email: jonathan@vidyo.com
Magnus Westerlund
Ericsson
Farogatan 6
SE-164 80 Kista
Sweden
Phone: +46 10 714 82 87
Email: magnus.westerlund@ericsson.com
Qin Wu
Huawei
101 Software Avenue, Yuhua District
Nanjing, Jiangsu 210012
China
Email: sunseawq@huawei.com
Colin Perkins
University of Glasgow
School of Computing Science
Glasgow G12 8QQ
United Kingdom
Email: csp@csperkins.org
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