draft-ietf-avtcore-multi-media-rtp-session-02.txt   draft-ietf-avtcore-multi-media-rtp-session-03.txt 
AVTCORE WG M. Westerlund AVTCORE WG M. Westerlund
Internet-Draft Ericsson Internet-Draft Ericsson
Updates: 3550, 3551 (if approved) C. Perkins Updates: 3550, 3551 (if approved) C. Perkins
Intended status: Standards Track University of Glasgow Intended status: Standards Track University of Glasgow
Expires: August 29, 2013 J. Lennox Expires: January 11, 2014 J. Lennox
Vidyo Vidyo
February 25, 2013 July 10, 2013
Multiple Media Types in an RTP Session Sending Multiple Types of Media in a Single RTP Session
draft-ietf-avtcore-multi-media-rtp-session-02 draft-ietf-avtcore-multi-media-rtp-session-03
Abstract Abstract
This document specifies how an RTP session can contain media streams This document specifies how an RTP session can contain media streams
with media from multiple media types such as audio, video, and text. with media from multiple media types such as audio, video, and text.
This has been restricted by the RTP Specification, and thus this This has been restricted by the RTP Specification, and thus this
document updates RFC 3550 and RFC 3551 to enable this behaviour for document updates RFC 3550 and RFC 3551 to enable this behaviour for
applications that satisfy the applicability for using multiple media applications that satisfy the applicability for using multiple media
types in a single RTP session. types in a single RTP session.
Status of this Memo Status of This Memo
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provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
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Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Motivation . . . . . . . . . . . . . . . . . . . . . . . . . . 4 3. Motivation . . . . . . . . . . . . . . . . . . . . . . . . . 4
3.1. NAT and Firewalls . . . . . . . . . . . . . . . . . . . . 4 3.1. NAT and Firewalls . . . . . . . . . . . . . . . . . . . . 4
3.2. No Transport Level QoS . . . . . . . . . . . . . . . . . . 5 3.2. No Transport Level QoS . . . . . . . . . . . . . . . . . 4
3.3. Architectural Equality . . . . . . . . . . . . . . . . . . 5 3.3. Architectural Equality . . . . . . . . . . . . . . . . . 5
4. Overview of Solution . . . . . . . . . . . . . . . . . . . . . 5 4. Overview of Solution . . . . . . . . . . . . . . . . . . . . 5
5. Applicability . . . . . . . . . . . . . . . . . . . . . . . . 6 5. Applicability . . . . . . . . . . . . . . . . . . . . . . . . 6
5.1. Usage of the RTP session . . . . . . . . . . . . . . . . . 6 5.1. Usage of the RTP session . . . . . . . . . . . . . . . . 6
5.2. Signalled Support . . . . . . . . . . . . . . . . . . . . 7 5.2. Signalled Support . . . . . . . . . . . . . . . . . . . . 7
5.3. Homogeneous Multi-party . . . . . . . . . . . . . . . . . 7 5.3. Homogeneous Multi-party . . . . . . . . . . . . . . . . . 7
5.4. Reduced number of Payload Types . . . . . . . . . . . . . 8 5.4. Reduced number of Payload Types . . . . . . . . . . . . . 8
5.5. Stream Differentiation . . . . . . . . . . . . . . . . . . 8 5.5. Stream Differentiation . . . . . . . . . . . . . . . . . 8
5.6. Non-compatible Extensions . . . . . . . . . . . . . . . . 9 5.6. Non-compatible Extensions . . . . . . . . . . . . . . . . 8
6. RTP Session Specification . . . . . . . . . . . . . . . . . . 9 6. RTP Session Specification . . . . . . . . . . . . . . . . . . 9
6.1. RTP Session . . . . . . . . . . . . . . . . . . . . . . . 10 6.1. RTP Session . . . . . . . . . . . . . . . . . . . . . . . 9
6.2. Sender Source Restrictions . . . . . . . . . . . . . . . . 12 6.2. Sender Source Restrictions . . . . . . . . . . . . . . . 11
6.3. Payload Type Applicability . . . . . . . . . . . . . . . . 12 6.3. Payload Type Applicability . . . . . . . . . . . . . . . 12
6.4. RTCP . . . . . . . . . . . . . . . . . . . . . . . . . . . 13 6.4. RTCP Considerations . . . . . . . . . . . . . . . . . . . 12
6.4.1. Timing out SSRCs . . . . . . . . . . . . . . . . . . . 14 7. Extension Considerations . . . . . . . . . . . . . . . . . . 12
6.4.2. Tuning RTCP transmissions . . . . . . . . . . . . . . 14 7.1. RTP Retransmission . . . . . . . . . . . . . . . . . . . 13
7. Extension Considerations . . . . . . . . . . . . . . . . . . . 17 7.2. Generic FEC . . . . . . . . . . . . . . . . . . . . . . . 13
7.1. RTP Retransmission . . . . . . . . . . . . . . . . . . . . 17 8. Signalling . . . . . . . . . . . . . . . . . . . . . . . . . 14
7.2. Generic FEC . . . . . . . . . . . . . . . . . . . . . . . 18 8.1. SDP-Based Signalling . . . . . . . . . . . . . . . . . . 14
8. Signalling . . . . . . . . . . . . . . . . . . . . . . . . . . 18 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 14
8.1. SDP-Based Signalling . . . . . . . . . . . . . . . . . . . 19 10. Security Considerations . . . . . . . . . . . . . . . . . . . 14
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 19 11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 15
10. Security Considerations . . . . . . . . . . . . . . . . . . . 19 12. References . . . . . . . . . . . . . . . . . . . . . . . . . 15
11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 19 12.1. Normative References . . . . . . . . . . . . . . . . . . 15
12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 20 12.2. Informative References . . . . . . . . . . . . . . . . . 16
12.1. Normative References . . . . . . . . . . . . . . . . . . . 20 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 17
12.2. Informative References . . . . . . . . . . . . . . . . . . 20
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 22
1. Introduction 1. Introduction
When the Real-time Transport Protocol (RTP) [RFC3550] was designed, When the Real-time Transport Protocol (RTP) [RFC3550] was designed,
close to 20 years ago, IP networks were very different compared to close to 20 years ago, IP networks were very different compared to
the ones in 2013 when this is written. The almost ubiquitous the ones in 2013 when this is written. The almost ubiquitous
deployment of Network Address Translators (NAT) and Firewalls has deployment of Network Address Translators (NAT) and Firewalls has
increased the cost and likely-hood of communication failure when increased the cost and likely-hood of communication failure when
using many different transport flows. Thus there exists a pressure using many different transport flows. Thus there exists a pressure
to reduce the number of concurrent transport flows. to reduce the number of concurrent transport flows.
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behaves as a single RTP stack entity it is classified as a single behaves as a single RTP stack entity it is classified as a single
endpoint. endpoint.
Media Stream: A sequence of RTP packets using a single SSRC that Media Stream: A sequence of RTP packets using a single SSRC that
together carries part or all of the content of a specific Media together carries part or all of the content of a specific Media
Type from a specific sender source within a given RTP session. Type from a specific sender source within a given RTP session.
Media Type: Audio, video, text or application whose form and meaning Media Type: Audio, video, text or application whose form and meaning
are defined by a specific real-time application. are defined by a specific real-time application.
QoS: Quality of Service, i.e. network mechanisms that intended to QoS: Quality of Service, i.e. network mechanisms that intended to
ensure that the packets within a flow or with a specific marking ensure that the packets within a flow or with a specific marking
are transported with certain properties. are transported with certain properties.
RTP Session: As defined by [RFC3550], the endpoints belonging to the RTP Session: As defined by [RFC3550], the endpoints belonging to the
same RTP Session are those that share a single SSRC space. That same RTP Session are those that share a single SSRC space. That
is, those endpoints can see an SSRC identifier transmitted by any is, those endpoints can see an SSRC identifier transmitted by any
one of the other endpoints. An endpoint can receive an SSRC one of the other endpoints. An endpoint can receive an SSRC
either as SSRC or as CSRC in RTP and RTCP packets. Thus, the RTP either as SSRC or as CSRC in RTP and RTCP packets. Thus, the RTP
Session scope is decided by the endpoints' network interconnection Session scope is decided by the endpoints' network interconnection
topology, in combination with RTP and RTCP forwarding strategies topology, in combination with RTP and RTCP forwarding strategies
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(i.e., it can switch between different audio codecs, since those are (i.e., it can switch between different audio codecs, since those are
both the same type of media, but cannot switch between audio and both the same type of media, but cannot switch between audio and
video). Different SSRCs MUST be used for the different media video). Different SSRCs MUST be used for the different media
sources, the same way multiple media sources of the same media type sources, the same way multiple media sources of the same media type
already have to do. The payload type will inform a receiver which already have to do. The payload type will inform a receiver which
media type the SSRC is being used for. Thus the payload type MUST be media type the SSRC is being used for. Thus the payload type MUST be
unique across all of the payload configurations independent of media unique across all of the payload configurations independent of media
type that is used in the RTP session. type that is used in the RTP session.
Some few extra considerations within the RTP sessions also needs to Some few extra considerations within the RTP sessions also needs to
be considered. RTCP bandwidth and regular reporting suppression be considered. RTCP bandwidth and regular reporting suppression (RTP
(RTP/AVPF and RTP/SAVPF) SHOULD be configured to reduce the impact /AVPF and RTP/SAVPF) SHOULD be configured to reduce the impact for
for bit-rate variations between streams and media types. It is also bit-rate variations between streams and media types. It is also
clarified how timeout calculations are to be done to avoid any clarified how timeout calculations are to be done to avoid any
issues. Certain payload types like FEC also need additional rules. issues. Certain payload types like FEC also need additional rules.
The final important part of the solution to this is to use signalling The final important part of the solution to this is to use signalling
and ensure that agreement on using multiple media types in an RTP and ensure that agreement on using multiple media types in an RTP
session exists, and how that then is configured. This memo describes session exists, and how that then is configured. This memo describes
some existing requirements, while an external reference defines how some existing requirements, while an external reference defines how
this is accomplished in SDP. this is accomplished in SDP.
5. Applicability 5. Applicability
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different cases. One case is where the RTP session contains multiple different cases. One case is where the RTP session contains multiple
participants in a common RTP session. This occurs for example in Any participants in a common RTP session. This occurs for example in Any
Source Multicast (ASM) and Transport Translator topologies as defined Source Multicast (ASM) and Transport Translator topologies as defined
in RTP Topologies [RFC5117]. It can also occur in some in RTP Topologies [RFC5117]. It can also occur in some
implementations of RTP mixers that share the same SSRC/CSRC space implementations of RTP mixers that share the same SSRC/CSRC space
across all participants. The second case is when the RTP session is across all participants. The second case is when the RTP session is
terminated in a middlebox and the other participants sources are terminated in a middlebox and the other participants sources are
projected or switched into each RTP session and rewritten on RTP projected or switched into each RTP session and rewritten on RTP
header level including SSRC mappings. header level including SSRC mappings.
For the first case, with a common RTP session or at least shared For the first case, with a common RTP session or at least shared SSRC
SSRC/CSRC values, all participants in multiparty communication are /CSRC values, all participants in multiparty communication are
REQUIRED to support multiple media types in an RTP session. An REQUIRED to support multiple media types in an RTP session. An
participant using two or more RTP sessions towards a multiparty participant using two or more RTP sessions towards a multiparty
session can't be collapsed into a single session with multiple media session can't be collapsed into a single session with multiple media
types. The reason is that in case of multiple RTP sessions, the same types. The reason is that in case of multiple RTP sessions, the same
SSRC value can be use in both RTP sessions without any issues, but SSRC value can be use in both RTP sessions without any issues, but
when collapsed to a single session there is an SSRC collision. In when collapsed to a single session there is an SSRC collision. In
addition some collisions can't be represented in the multiple addition some collisions can't be represented in the multiple
separate RTP sessions. For example, in a session with audio and separate RTP sessions. For example, in a session with audio and
video, an SSRC value used for video will not show up in the Audio RTP video, an SSRC value used for video will not show up in the Audio RTP
session at the participant using multiple RTP sessions, and thus not session at the participant using multiple RTP sessions, and thus not
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each SSRC will be associated with a specific media type, communicated each SSRC will be associated with a specific media type, communicated
through the RTP payload type, allowing a middlebox to do media type through the RTP payload type, allowing a middlebox to do media type
specific operations. The second argument is that in many contexts specific operations. The second argument is that in many contexts
blind combining without additional contexts are anyway not suitable. blind combining without additional contexts are anyway not suitable.
Regarding bullet 5 this is a understood and explicitly stated Regarding bullet 5 this is a understood and explicitly stated
applicability limitations for the method described in this document. applicability limitations for the method described in this document.
6.2. Sender Source Restrictions 6.2. Sender Source Restrictions
A SSRC in the RTP session MUST only send one media type (audio, A SSRC in the RTP session MUST only send one media type (audio,
video, text etc.) during the SSRC's lifetime. The main motivation is video, text etc.) during the SSRC's lifetime. The main motivation
that a given SSRC has its own RTP timestamp and sequence number is that a given SSRC has its own RTP timestamp and sequence number
spaces. The same way that you can't send two streams of encoded spaces. The same way that you can't send two streams of encoded
audio on the same SSRC, you can't send one audio and one video audio on the same SSRC, you can't send one audio and one video
encoding on the same SSRC. Each media encoding when made into an RTP encoding on the same SSRC. Each media encoding when made into an RTP
stream needs to have the sole control over the sequence number and stream needs to have the sole control over the sequence number and
timestamp space. If not, one would not be able to detect packet loss timestamp space. If not, one would not be able to detect packet loss
for that particular stream. Nor can one easily determine which clock for that particular stream. Nor can one easily determine which clock
rate a particular SSRCs timestamp will increase with. For additional rate a particular SSRCs timestamp will increase with. For additional
arguments why RTP payload type based multiplexing of multiple media arguments why RTP payload type based multiplexing of multiple media
streams doesn't work see Appendix A in streams doesn't work see Appendix A in
[I-D.westerlund-avtcore-multiplex-architecture]. [I-D.westerlund-avtcore-multiplex-architecture].
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types. In fact this document (Section 7.2) suggest that for usage of types. In fact this document (Section 7.2) suggest that for usage of
Generic FEC (XOR-based) as defined in RFC 5109 can actually use a Generic FEC (XOR-based) as defined in RFC 5109 can actually use a
single media type when used with independent RTP sessions for source single media type when used with independent RTP sessions for source
and repair data. and repair data.
Note a particular SSRC carrying Generic FEC will clearly only Note a particular SSRC carrying Generic FEC will clearly only
protect a specific SSRC and thus that instance is bound to the protect a specific SSRC and thus that instance is bound to the
SSRC's media type. For this specific case, it is possible to have SSRC's media type. For this specific case, it is possible to have
one be applicable to both. However, in cases when the signalling one be applicable to both. However, in cases when the signalling
is setup to enable fall back to using separate RTP sessions, then is setup to enable fall back to using separate RTP sessions, then
using a different media type, e.g. application, than the media using a different media type, e.g. application, than the media
being protected can create issues. being protected can create issues.
6.4. RTCP 6.4. RTCP Considerations
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.
When it comes to handling multiple SSRCs in an RTP session there is a
clarification under discussion in Real-Time Transport Protocol (RTP)
Considerations for Multi-Stream Endpoints
[I-D.lennox-avtcore-rtp-multi-stream]. When it comes to configuring
RTCP the need for regular periodic reporting needs to be weighted
against any feedback or control messages being sent. The
applications using RTP/AVPF or RTP/SAVPF are RECOMMENDED to consider
setting trr-int parameter to a value suitable for the applications
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 used 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 an messages/packet and only use those that apply to
that particular SSRC and its media type. Signalling solutions might
have shortcomings when it comes to indicate that a particular set of
RTCP reports or feedback messages only apply to a particular media
type within an RTP session.
6.4.1. Timing out SSRCs
All used SSRCs in the 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 such issues when using this specification we make
clarification on the calculations.
For RTP/AVP, RTP/SAVP, RTP/AVPF, and RTP/SAVPF with T_rr_interval = 0
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. If using either the RTP/AVPF or RTP/SAVPF profiles with
T_rr_interval != 0 then the calculation as specified in Section 3.5.4
of RFC 4585 SHALL be used with a multiplier of 5, i.e. Tmin in the
Td calculation is the T_rr_interval.
Note: 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/AVP endpoints
will prematurely timeout the RTP/AVPF endpoints due to their
different RTCP timeout intervals. Since an RTP session can only use
a single RTP profile, this issue ought never occur. If such mixed
RTP profiles are used, however, the RTP/AVPF session MUST NOT use a
non-zero T_rr_interval that is smaller than 5 seconds.
(tbd: it has been suggested that a minimum non-zero T_rr_interval of
4 seconds is more appropriate, due to the nature of the timing
rules).
6.4.2. Tuning RTCP transmissions
This sub-section discusses what tuning can be done to reduce
downsides of the shared RTCP packet intervals.
When using the RTP/AVP or RTP/SAVP profile the tuning one can do is
very limited. The controls one has are very limited to the RTCP
bandwidth values and if one scales the minimum RTCP interval
according to the bandwidth. As the scheduling algorithm includes
both random factors and reconsideration, one can't simply calculate
the expected average transmission interval using 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 are larger than the
ratio of RTCP bandwidth for senders in relation to the total RTCP
bandwidth, then senders and receivers are treated together.
Lets start with some basic observations:
a. Unless 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 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. A RTP Session
bandwidth of 72 kbps results in Tmin being 5 seconds. A 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 smaller Tmin, and hence
more frequent RTCP report, as discussed below).
c. Lets 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, a SDES item with at least a CNAME (17
bytes value) 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, it is an exponential increasing one. 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.
To conclude, with RTP/AVP and RTP/SAVP the key limitation for small
unicast sessions are going to be the Tmin value. Thus the RTP
session bandwidth configured in RTCP has to be sufficient high to
reach the reporting goals the application has following the rules for
scaled minimal RTCP interval.
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 and regular reporting interval is instead
affected of the regular bandwidth based calculation and the
T_rr_interval. This has the affect 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 is the
governing factors. Now it also becomes important to separate between
the applications 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 a hard as no regular RTCP packet can be
sent less than T_rr_interval after the previous regular packet
packet.
So for applications that has a use for feedback packets for some
media streams, for example video packets but don't want to 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. 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
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 one is more likely to affect the RTCP behaviour and
performance, than when using RTP/AVP, as their is fewer limitations
affecting the RTCP transmission.
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 T_rr_interval of 0 or with another low value,
which will be significantly lower than Td still has its 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 exist no method within the specification for using different Guidelines for handling RTCP when sending multiple media streams with
regular RTCP reporting interval depending on media type or individual disparate rates in a single RTP session are outlined in
media stream. [I-D.ietf-avtcore-rtp-multi-stream]. These guidelines apply when
sending multiple types of media in a single RTP session if the
different types of media have different rates.
7. Extension Considerations 7. Extension Considerations
This section discusses the impact on some RTP/RTCP extensions due to This section discusses the impact on some RTP/RTCP extensions due to
usage of multiple media types in on RTP session. Only extensions usage of multiple media types in on RTP session. Only extensions
where something worth noting has been included. where something worth noting has been included.
7.1. RTP Retransmission 7.1. RTP Retransmission
SSRC-multiplexed RTP retransmission [RFC4588] is actually very SSRC-multiplexed RTP retransmission [RFC4588] is actually very
skipping to change at page 18, line 12 skipping to change at page 13, line 37
contradict both the FID semantics [RFC5888] and an assumption in the contradict both the FID semantics [RFC5888] and an assumption in the
RTP retransmission specification [RFC4588]. RTP retransmission specification [RFC4588].
7.2. Generic FEC 7.2. Generic FEC
The RTP Payload Format for Generic Forward Error Correction The RTP Payload Format for Generic Forward Error Correction
[RFC5109], and also its predecessor [RFC2733], requires some [RFC5109], and also its predecessor [RFC2733], requires some
considerations, and they are different depending on what type of considerations, and they are different depending on what type of
configuration of usage one has. configuration of usage one has.
Independent RTP Sessions, i.e. where source and repair data are sent Independent RTP Sessions, i.e. where source and repair data are sent
in different RTP sessions. As this mode of configuration requires in different RTP sessions. As this mode of configuration requires
different RTP session, there has to be at least one RTP session for different RTP session, there has to be at least one RTP session for
source data, this session can be one using multiple media types. The source data, this session can be one using multiple media types. The
repair session only needs one RTP Payload type indicating repair repair session only needs one RTP Payload type indicating repair
data, i.e. x/ulpfec or x/parityfec depending if RFC 5109 or RFC 2733 data, i.e. x/ulpfec or x/parityfec depending if RFC 5109 or RFC 2733
is used. The media type in this session is not relevant and can in is used. The media type in this session is not relevant and can in
theory be any of the defined ones. It is RECOMMENDED that one uses theory be any of the defined ones. It is RECOMMENDED that one uses
"Application". "Application".
In stream, using RTP Payload for Redundant Audio Data [RFC2198] In stream, using RTP Payload for Redundant Audio Data [RFC2198]
combining repair and source data in the same packets. This is combining repair and source data in the same packets. This is
possible to use within a single RTP session. However, the usage and possible to use within a single RTP session. However, the usage and
configuration of the payload types can create an issue. First of all configuration of the payload types can create an issue. First of all
it might be necessary to have one payload type per media type for the it might be necessary to have one payload type per media type for the
FEC repair data payload format, i.e. one for audio/ulpfec and one for FEC repair data payload format, i.e. one for audio/ulpfec and one
text/ulpfec if audio and text are combined in an RTP session. for text/ulpfec if audio and text are combined in an RTP session.
Secondly each combination of source payload and its FEC repair data Secondly each combination of source payload and its FEC repair data
has to be an explicit configured payload type. This has potential has to be an explicit configured payload type. This has potential
for making the limitation of RTP payload types available into a real for making the limitation of RTP payload types available into a real
issue. issue.
8. Signalling 8. Signalling
The Signalling requirements The Signalling requirements
Establishing an RTP session with multiple media types requires Establishing an RTP session with multiple media types requires
skipping to change at page 20, line 9 skipping to change at page 15, line 31
11. Acknowledgements 11. Acknowledgements
The authors would like to thank Christer Holmberg, Gunnar Hellstroem, The authors would like to thank Christer Holmberg, Gunnar Hellstroem,
and Charles Eckel for the feedback on the document. and Charles Eckel for the feedback on the document.
12. References 12. References
12.1. Normative References 12.1. Normative References
[I-D.ietf-avtcore-rtp-multi-stream]
Lennox, J., Westerlund, M., Wu, W., and C. Perkins, "RTP
Considerations for Endpoints Sending Multiple Media
Streams", draft-ietf-avtcore-rtp-multi-stream-00 (work in
progress), April 2013.
[I-D.ietf-mmusic-sdp-bundle-negotiation] [I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings, Holmberg, C., Alvestrand, H., and C. Jennings,
"Multiplexing Negotiation Using Session Description "Multiplexing Negotiation Using Session Description
Protocol (SDP) Port Numbers", Protocol (SDP) Port Numbers", draft-ietf-mmusic-sdp-
draft-ietf-mmusic-sdp-bundle-negotiation-03 (work in bundle-negotiation-04 (work in progress), June 2013.
progress), February 2013.
[I-D.lennox-avtcore-rtp-multi-stream]
Lennox, J. and M. Westerlund, "Real-Time Transport
Protocol (RTP) Considerations for Endpoints Sending
Multiple Media Streams",
draft-lennox-avtcore-rtp-multi-stream-01 (work in
progress), October 2012.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997. Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003. Applications", STD 64, RFC 3550, July 2003.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551, Video Conferences with Minimal Control", STD 65, RFC 3551,
skipping to change at page 20, line 43 skipping to change at page 16, line 18
12.2. Informative References 12.2. Informative References
[I-D.lennox-payload-ulp-ssrc-mux] [I-D.lennox-payload-ulp-ssrc-mux]
Lennox, J., "Supporting Source-Multiplexing of the Real- Lennox, J., "Supporting Source-Multiplexing of the Real-
Time Transport Protocol (RTP) Payload for Generic Forward Time Transport Protocol (RTP) Payload for Generic Forward
Error Correction", draft-lennox-payload-ulp-ssrc-mux-00 Error Correction", draft-lennox-payload-ulp-ssrc-mux-00
(work in progress), February 2013. (work in progress), February 2013.
[I-D.westerlund-avtcore-multiplex-architecture] [I-D.westerlund-avtcore-multiplex-architecture]
Westerlund, M., Burman, B., Perkins, C., and H. Westerlund, M., Perkins, C., and H. Alvestrand,
Alvestrand, "Guidelines for using the Multiplexing "Guidelines for using the Multiplexing Features of RTP",
Features of RTP", draft-westerlund-avtcore-multiplex-architecture-03 (work
draft-westerlund-avtcore-multiplex-architecture-02 (work in progress), February 2013.
in progress), July 2012.
[I-D.westerlund-avtcore-transport-multiplexing] [I-D.westerlund-avtcore-transport-multiplexing]
Westerlund, M. and C. Perkins, "Multiple RTP Sessions on a Westerlund, M. and C. Perkins, "Multiple RTP Sessions on a
Single Lower-Layer Transport", Single Lower-Layer Transport", draft-westerlund-avtcore-
draft-westerlund-avtcore-transport-multiplexing-04 (work transport-multiplexing-05 (work in progress), February
in progress), October 2012. 2013.
[RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., [RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse- Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
Parisis, "RTP Payload for Redundant Audio Data", RFC 2198, Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
September 1997. September 1997.
[RFC2733] Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format [RFC2733] Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format
for Generic Forward Error Correction", RFC 2733, for Generic Forward Error Correction", RFC 2733, December
December 1999. 1999.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006. Description Protocol", RFC 4566, July 2006.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control "Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July
July 2006. 2006.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R. [RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588, Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
July 2006. July 2006.
[RFC5109] Li, A., "RTP Payload Format for Generic Forward Error [RFC5109] Li, A., "RTP Payload Format for Generic Forward Error
Correction", RFC 5109, December 2007. Correction", RFC 5109, December 2007.
[RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117, [RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117,
January 2008. January 2008.
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