6. Adding and Removing SSRCs
The set of SSRCs present in a single RTP session can vary over time
due to changes in the number of endpoints in the session or due to
changes in the number or type of RTP streams being sent.
Every endpoint in an RTP session will have at least one SSRC that it
uses for RTCP reporting, and for sending media if desired. It can
also have additional SSRCs, for sending extra media sources or for
additional RTCP reporting. If the set of media sources being sent
changes, then the set of SSRCs being sent will change. Changes in
the media format or clock rate might also require changes in the set
of SSRCs used. An endpoint can also have more SSRCs than it has
active RTP streams, and send RTCP relating to SSRCs that are not
currently sending RTP data packets so that its peers are aware of the
SSRCs, and have the associated context (e.g., clock synchronization
and an SDES CNAME) in place to be able to play out media as soon as
they becomes active.
In the following, we describe some considerations around adding and
removing RTP streams and their associated SSRCs.
6.1. Adding RTP Streams
When an endpoint joins an RTP session, it can have zero, one, or more
RTP streams it will send, or that it is prepared to send. If it has
no RTP stream it plans to send, it still needs an SSRC that will be
used to send RTCP feedback. If it will send one or more RTP streams,
it will need the corresponding number of SSRC values. The SSRCs used
by an endpoint are made known to other endpoints in the RTP session
by sending RTP and RTCP packets. SSRCs can also be signaled using
non-RTP means (e.g., [RFC5576]). Unless restricted by signaling, an
endpoint can, at any time, send an additional RTP stream, identified
by a new SSRC (this might be associated with a signaling event, but
that is outside the scope of this memo). This makes the new SSRC
visible to the other endpoints in the session, since they share the
single SSRC space inherent in the definition of an RTP session.
An endpoint that has never sent an RTP stream will have an SSRC that
it uses for RTCP reporting. If that endpoint wants to start sending
an RTP stream, it is RECOMMENDED that it use its existing SSRC for
that stream, since otherwise the participant count in the RTP session
will be unnecessarily increased, leading to a longer RTCP reporting
interval and larger RTCP reports due to cross reporting. If the
endpoint wants to start sending more than one RTP stream, it will
need to generate a new SSRC for the second and any subsequent RTP
An endpoint that has previously stopped sending an RTP stream, and
that wants to start sending a new RTP stream, cannot generally reuse
the existing SSRC, and often needs to generate a new SSRC, because an
SSRC cannot change media type (e.g., audio to video) or RTP timestamp
clock rate [RFC7160] and because the SSRC might be associated with a
particular semantic by the application (note: an RTP stream can pause
and restart using the same SSRC, provided RTCP is sent for that SSRC
during the pause; these rules only apply to new RTP streams reusing
an existing SSRC).
6.2. Removing RTP Streams
An SSRC is removed from an RTP session in one of two ways. When an
endpoint stops sending RTP and RTCP packets using an SSRC, then that
SSRC will eventually time out as described in Section 6.3.5 of
[RFC3550]. Alternatively, an SSRC can be explicitly removed from use
by sending an RTCP BYE packet as described in Section 6.3.7 of
[RFC3550]. It is RECOMMENDED that SSRCs be removed from use by
sending an RTCP BYE packet. Note that [RFC3550] requires that the
RTCP BYE SHOULD be the last RTP/RTCP packet sent in the RTP session
for an SSRC. If an endpoint needs to restart an RTP stream after
sending an RTCP BYE for its SSRC, it needs to generate a new SSRC
value for that stream.
The finality of sending RTCP BYE means that endpoints need to
consider if the ceasing of transmission of an RTP stream is temporary
or permanent. Temporary suspension of media transmission using a
particular RTP stream (SSRC) needs to maintain that SSRC as an active
participant, by continuing RTCP transmission for it. That way the
media sending can be resumed immediately, knowing that the context is
in place. When permanently halting transmission, a participant needs
to send an RTCP BYE to allow the other participants to use the RTCP
bandwidth resources and clean up their state databases.
An endpoint that ceases transmission of all its RTP streams but
remains in the RTP session MUST maintain at least one SSRC that is to
be used for RTCP reporting and feedback (i.e., it cannot send a BYE
for all SSRCs, but needs to retain at least one active SSRC). As
some Feedback packets can be bound to media type, there might be a
need to maintain one SSRC per media type within an RTP session. An
alternative can be to create a new SSRC to use for RTCP reporting and
feedback. However, to avoid the perception that an endpoint drops
completely out of an RTP session, such a new SSRC ought to be
established first -- before terminating all the existing SSRCs.
7. RTCP Considerations for Streams with Disparate Rates
An RTP session has a single set of parameters that configure the
session bandwidth. These are the RTCP sender and receiver fractions
(e.g., the SDP "b=RR:" and "b=RS:" lines [RFC3556]) 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 base RTCP reporting interval, before randomization,
will be the same for every sending SSRC in an RTP session.
Similarly, every receiving SSRC in an RTP session will have the same
base reporting interval, although this can differ from the reporting
interval chosen by sending SSRCs. This uniform RTCP reporting
interval for all SSRCs can result in RTCP reports being sent more
often, or too seldom, than is considered desirable for an RTP stream.
For example, consider a scenario in which an audio flow sending at
tens of kilobits per second is multiplexed into an RTP session with a
multi-megabit high-quality video flow. If the session bandwidth is
configured based on the video sending rate, and the default RTCP
bandwidth fraction of 5% of the session bandwidth is used, it is
likely that the RTCP bandwidth will exceed the audio sending rate.
If the reduced minimum RTCP interval described in Section 6.2 of
[RFC3550] is then used in the session, as appropriate for video where
rapid feedback on damaged I-frames is wanted, the uniform reporting
interval for all senders could mean that audio sources are expected
to send RTCP packets more often than they send audio data packets.
This bandwidth mismatch can be reduced by careful tuning of the RTCP
parameters, especially trr_int when the RTP/AVPF profile is used, but
cannot be avoided entirely as it is inherent in the design of the
RTCP timing rules, and affects all RTP sessions that contain flows
with greatly mismatched bandwidth.
Different media rates or desired RTCP behaviors can also occur with
SSRCs carrying the same media type. A common case in multiparty
conferencing is when a small number of video streams are shown in
high resolution, while the others are shown as low-resolution
thumbnails, with the choice of which is shown in high resolution
being voice-activity controlled. Here the differences are both in
actual media rate and in choices for what feedback messages might be
needed. Other examples of differences that can exist are due to the
intended usage of a media source. A media source carrying the video
of the speaker in a conference is different from a document camera.
Basic parameters that can differ in this case are frame-rate,
acceptable end-to-end delay, and the Signal-to-Noise Ratio (SNR)
fidelity of the image. These differences affect not only the needed
bitrates, but also possible transmission behaviors, usable repair
mechanisms, what feedback messages the control and repair requires,
the transmission requirements on those feedback messages, and
monitoring of the RTP stream delivery. Other similar scenarios can
Sending multiple media types in a single RTP session causes that
session to contain more SSRCs than if each media type was sent in a
separate RTP session. For example, if two participants each send an
audio and a video RTP stream in a single RTP session, that session
will comprise four SSRCs; but if separate RTP sessions had been used
for audio and video, each of those two RTP sessions would comprise
only two SSRCs. Hence, sending multiple RTP streams in an RTP
session increases the amount of cross reporting between the SSRCs, as
each SSRC reports on all other SSRCs in the session. This increases
the size of the RTCP reports, causing them to be sent less often than
would be the case if separate RTP sessions where used for a given
Finally, when an RTP session contains multiple media types, it is
important to note that the RTCP reception quality reports, feedback
messages, and extended report blocks used might not be applicable to
all media types. Endpoints will need to consider the media type of
each SSRC, and only send or process reports and feedback that apply
to that particular SSRC and its media type. Signaling 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.
From an RTCP perspective, therefore, it can be seen that there are
advantages to using separate RTP sessions for each media source,
rather than sending multiple media sources in a single RTP session.
However, these are frequently offset by the need to reduce port use,
to ease NAT/firewall traversal, achieved by combining media sources
into a single RTP session. The following sections consider some of
the issues with using RTCP in sessions with multiple media sources in
7.1. Timing Out SSRCs
Various issues have been identified with timing out SSRC values when
sending multiple RTP streams in an RTP session.
7.1.1. Problems with the RTP/AVPF T_rr_interval Parameter
The RTP/AVPF profile includes a method to prevent regular RTCP
reports from being sent too often. This mechanism is described in
Section 3.5.3 of [RFC4585]; it is controlled by the T_rr_interval
parameter. It works as follows. When a regular RTCP report is sent,
a new random value, T_rr_current_interval, is generated, drawn evenly
in the range 0.5 to 1.5 times T_rr_interval. If a regular RTCP
packet is to be sent earlier than T_rr_current_interval seconds after
the previous regular RTCP packet, and there are no feedback messages
to be sent, then that regular RTCP packet is suppressed and the next
regular RTCP packet is scheduled. The T_rr_current_interval is
recalculated each time a regular RTCP packet is sent. The benefit of
suppression 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 skews the distribution of
the RTCP sending intervals compared to the regular RTCP reporting
intervals. The standard RTCP timing rules, including reconsideration
and the compensation factor, result in the intervals between sending
RTCP packets having a distribution that is skewed towards the upper
end of the range [0.5/1.21828, 1.5/1.21828]*Td, where Td is the
deterministic calculated RTCP reporting interval. With Td = 5 s,
this distribution covers the range [2.052 s, 6.156 s]. In
comparison, the RTP/AVPF suppression rules act in an interval that is
0.5 to 1.5 times T_rr_interval; for T_rr_interval = 5s, this is
[2.5 s, 7.5 s].
The effect of this is that the time between consecutive RTCP packets
when using T_rr_interval suppression can become large. The maximum
time interval between sending one regular RTCP packet and the next,
when T_rr_interval is being used, occurs when T_rr_current_interval
takes its maximum value and a regular RTCP packet is suppressed at
the end of the suppression period, then the next regular RTCP packet
is scheduled after its largest possible reporting interval. Taking
the worst case of the two intervals gives a maximum time between two
RTCP reports of 1.5*T_rr_interval + 1.5/1.21828*Td.
This behavior can be surprising when Td and T_rr_interval have the
same value. That is, when T_rr_interval is configured to match the
regular RTCP reporting interval. In this case, one might expect that
regular RTCP packets are sent according to their usual schedule, but
feedback packets can be sent early. However, the above-mentioned
issue results in the RTCP packets actually being sent in the range
[0.5*Td, 2.731*Td] with a highly non-uniform distribution, rather
than the range [0.41*Td, 1.23*Td]. This is perhaps unexpected, but
is not a problem in itself. However, when coupled with packet loss,
it raises the issue of premature timeout.
7.1.2. Avoiding Premature Timeout
In RTP/AVP [RFC3550] the timeout behavior is simple; it is 5 times
Td, where Td is calculated with a Tmin value of 5 seconds. In other
words, if the configured RTCP bandwidth allows for an average RTCP
reporting interval shorter than 5 seconds, the timeout is 25 seconds
of no activity from the SSRC (RTP or RTCP); otherwise, the timeout is
5 average reporting intervals.
RTP/AVPF [RFC4585] introduces different timeout behaviors depending
on the value of T_rr_interval. When T_rr_interval is 0, it uses the
same timeout calculation as RTP/AVP. However, when T_rr_interval is
non-zero, it replaces Tmin in the timeout calculation, most likely to
speed up detection of timed out SSRCs. However, using a non-zero
T_rr_interval has two consequences for RTP behavior.
First, due to suppression, the number of RTP and RTCP packets sent by
an SSRC that is not an active RTP sender can become very low, because
of the issue discussed in Section 7.1.1. As the RTCP packet interval
can be as long as 2.73*Td, during a 5*Td time period, an endpoint
might in fact transmit only a single RTCP packet. The long intervals
result in fewer RTCP packets, to a point where a single RTCP packet
loss can sometimes result in timing out an SSRC.
Second, the RTP/AVPF changes to the timeout rules reduce robustness
to misconfiguration. It is common to use RTP/AVPF configured such
that RTCP packets can be sent frequently to allow rapid feedback;
however, this makes timeouts very sensitive to T_rr_interval. For
example, if two SSRCs are configured, one with T_rr_interval = 0.1 s
and the other with T_rr_interval = 0.6 s, then this small difference
will result in the SSRC with the shorter T_rr_interval timing out the
other if it stops sending RTP packets, since the other RTCP reporting
interval is more than five times its own. When RTP/AVP is used, or
RTP/AVPF with T_rr_interval = 0, this is a non-issue, as the timeout
period will be 25 s, and differences between configured RTCP
bandwidth can only cause premature timeouts when the reporting
intervals are greater than 5 s and differ by a factor of five. To
limit the scope for such problematic misconfiguration, we define an
update to the RTP/AVPF timeout rules in Section 7.1.4.
7.1.3. Interoperability between RTP/AVP and RTP/AVPF
If endpoints implementing the RTP/AVP and RTP/AVPF profiles (or their
secure variants) are combined within a single RTP session, and the
RTP/AVPF endpoints use a non-zero T_rr_interval that is significantly
below 5 seconds, there is a risk that the RTP/AVPF endpoints will
prematurely time out the SSRCs of the RTP/AVP endpoints, due to their
different RTCP timeout rules. Conversely, if the RTP/AVPF endpoints
use a T_rr_interval that is significantly larger than 5 seconds,
there is a risk that the RTP/AVP endpoints will time out the SSRCs of
the RTP/AVPF endpoints.
Mixing endpoints using two different RTP profiles within a single RTP
session is NOT RECOMMENDED. However, if mixed RTP profiles are used,
and the RTP/AVPF endpoints are not updated to follow Section 7.1.4 of
this memo, then the RTP/AVPF session SHOULD be configured to use
T_rr_interval = 4 seconds to avoid premature timeouts.
The choice of T_rr_interval = 4 seconds for interoperability might
appear strange. Intuitively, this value ought to be 5 seconds, to
make both the RTP/AVP and RTP/AVPF use the same timeout period.
However, the behavior outlined in Section 7.1.1 shows that actual
RTP/AVPF reporting intervals can be longer than expected. Setting
T_rr_interval = 4 seconds gives actual RTCP intervals near to those
expected by RTP/AVP, ensuring interoperability.
7.1.4. Updated SSRC Timeout Rules
To ensure interoperability and avoid premature timeouts, all SSRCs in
an RTP session MUST use the same timeout behavior. However, previous
specifications are inconsistent in this regard. To avoid
interoperability issues, this memo updates the timeout rules as
o For the RTP/AVP, RTP/SAVP, RTP/AVPF, and RTP/SAVPF profiles, the
timeout interval SHALL be calculated using a multiplier of five
times the deterministic RTCP reporting interval. That is, the
timeout interval SHALL be 5*Td.
o For the RTP/AVP, RTP/SAVP, RTP/AVPF, and RTP/SAVPF profiles,
calculation of Td, for the purpose of calculating the participant
timeout only, SHALL be done using a Tmin value of 5 seconds and
not the reduced minimal interval, even if the reduced minimum
interval is used to calculate RTCP packet transmission intervals.
This changes the behavior for the RTP/AVPF or RTP/SAVPF profiles when
T_rr_interval != 0. Specifically, the first paragraph of
Section 3.5.4 of [RFC4585] is updated to use Tmin instead of
T_rr_interval in the timeout calculation for RTP/AVPF entities.
7.2. Tuning RTCP Transmissions
This subsection discusses what tuning can be done to reduce the
downsides of the shared RTCP packet intervals. First, what
possibilities exist for the RTP/AVP [RFC3551] profile are listed
followed by what additional tools are provided by RTP/AVPF [RFC4585].
7.2.1. RTP/AVP and RTP/SAVP
When using the RTP/AVP or RTP/SAVP profiles, the options for tuning
the RTCP reporting intervals are limited to the RTCP sender and
receiver bandwidth, and whether the minimum RTCP interval is scaled
according to the bandwidth. As the scheduling algorithm includes
both randomization and reconsideration, one cannot simply calculate
the expected average transmission interval using the formula for Td
given in Section 6.3.1 of [RFC3550]. However, by considering the
inputs to that expression, and the randomization and reconsideration
rules, we can begin to understand the behavior of the RTCP
Let's start with some basic observations:
a. Unless the scaled minimum RTCP interval is used, Td prior to
randomization and reconsideration can never be less than Tmin.
The default value of Tmin is 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 per second.
In SDP, the RTP session bandwidth is signaled using a "b=AS"
line. 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 frame-per-second video stream requires an RTP
session bandwidth of 9 Mbps. Use of the RTP/AVPF or RTP/SAVPF
profile allows more frequent RTCP reports for the same bandwidth,
as discussed below.
c. The value of Td scales with the number of SSRCs and the average
size of the RTCP reports to keep the overall RTCP bandwidth
d. The actual transmission interval for a Td value is in the range
[0.5*Td/1.21828, 1.5*Td/1.21828], and the distribution is skewed,
due to reconsideration, with the majority of the probability mass
being above Td. This means, for example, that for Td = 5 s, the
actual transmission interval will be distributed in the range
[2.052 s, 6.156 s], and tending towards the upper half of the
interval. Note that Tmin parameter limits the value of Td before
randomization and reconsideration are applied, so the actual
transmission interval will cover a range extending below Tmin.
Given the above, we can calculate the number of SSRCs, n, that an RTP
session with 5% of the session bandwidth assigned to RTCP can support
while maintaining Td equal to Tmin. This will tell us how many RTP
streams we can report on, keeping the RTCP overhead within acceptable
bounds. We make two assumptions that simplify the calculation: that
all SSRCs are senders, and that they all send compound RTCP packets
comprising an SR packet with n-1 report blocks, followed by an SDES
packet containing a 16 octet CNAME value [RFC7022] (such RTCP packets
will vary in size between 54 and 798 octets depending on n, up to the
maximum of 31 report blocks that can be included in an SR packet).
If we put this packet size, and a 5% RTCP bandwidth fraction into the
RTCP interval calculation in Section 6.3.1 of [RFC3550], and
calculate the value of n needed to give Td = Tmin for the scaled
minimum interval, we find n=9 SSRCs can be supported (irrespective of
the interval, due to the way the reporting interval scales with the
session bandwidth). We see that to support more SSRCs without
changing the scaled minimum interval, we need to increase the RTCP
bandwidth fraction from 5%; changing the session bandwidth to a
higher value would reduce the Tmin. However, if using the default 5%
allocation of RTCP bandwidth, an increase will result in more SSRCs
being supported given a fixed Td target.
Based on the above, when using the RTP/AVP profile or the RTP/SAVP
profile, the key limitation for rapid RTCP reporting in small unicast
sessions is going to be the Tmin value. 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
7.2.2. RTP/AVPF and RTP/SAVPF
When using RTP/AVPF or RTP/SAVPF, we have a powerful additional tool
for tuning RTCP transmissions: the T_rr_interval parameter. Use of
this parameter allows short RTCP reporting intervals; alternatively
it gives the ability to sent frequent RTCP feedback without sending
frequent regular RTCP reports.
The use of the RTP/AVPF or RTP/SAVPF profile with T_rr_interval set
to a value greater than zero but smaller than Tmin allows more
frequent RTCP feedback than the RTP/AVP or RTP/SAVP profiles, for a
given RTCP bandwidth. This happens because Tmin is set to zero after
the transmission of the initial RTCP report, causing the reporting
interval for later packet to be determined by the usual RTCP
bandwidth-based calculation, with Tmin=0, and the T_rr_interval.
This has the effect that we are no longer restricted by the minimal
interval (whether the default 5-second minimum or the reduced minimum
interval). Rather, the RTCP bandwidth and the T_rr_interval are the
governing factors, allowing faster feedback. Applications that care
about rapid regular RTCP feedback ought to consider using the RTP/
AVPF or RTP/SAVPF profile, even if they don't use the feedback
features of that profile.
The use of the RTP/AVPF or RTP/SAVPF profile allows RTCP feedback
packets to be sent frequently, without also requiring regular RTCP
reports to be sent frequently, since T_rr_interval limits the rate at
which regular RTCP packets can be sent, while still permitting RTCP
feedback packets to be sent. Applications that can use feedback
packets for some RTP streams, e.g., video streams, but don't want
frequent regular reporting for other RTP streams, can 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,
for the video reporting. This allows the available RTCP bandwidth to
be devoted on the feedback that provides the most utility for the
Using T_rr_interval still requires one to determine suitable values
for the RTCP bandwidth value. Indeed, it might make this choice even
more important, as this is more likely to affect the RTCP behavior
and performance than when using the RTP/AVP or RTP/SAVP profile, as
there are fewer limitations affecting the RTCP transmission.
When T_rr_interval is non-zero, there are configurations that need to
be avoided. If the RTCP bandwidth chosen is such that the Td value
is smaller than, but close to, T_rr_interval, then the actual regular
RTCP packet transmission interval can become very large, as discussed
in Section 7.1.1. 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
the range becoming [0.5*T_rr_interval, 1.81*T_rr_interval].
With the RTP/AVPF or RTP/SAVPF profiles, using T_rr_interval = 0 has
utility and results in a behavior where the RTCP transmission is only
limited by the bandwidth, i.e., no Tmin limitations at all. This
allows more frequent regular RTCP reporting than can be achieved
using the RTP/AVP profile. Many configurations of RTCP will not
consume all the bandwidth that they have been configured to use, but
this configuration will consume what it has been given. Note that
the same behavior will be achieved as long as T_rr_interval is
smaller than 1/3 of Td as that prevents T_rr_interval from affecting
There exists no method for using different regular RTCP reporting
intervals depending on the media type or individual RTP stream, other
than using a separate RTP session for each type or stream.
8. Security Considerations
When using the secure RTP protocol (RTP/SAVP) [RFC3711], or the
secure variant of the feedback profile (RTP/SAVPF) [RFC5124], the
cryptographic context of a compound secure RTCP 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] that
allow use of per-SSRC keying.
Otherwise, the standard security considerations of RTP apply; sending
multiple RTP streams from a single endpoint in a single RTP session
does not appear to have different security consequences than sending
the same number of RTP streams spread across different RTP sessions.
The authors like to thank Harald Alvestrand and everyone else who has
been involved in the development of this document.
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