Last Call Review of draft-ietf-avtcore-multi-media-rtp-session-11
review-ietf-avtcore-multi-media-rtp-session-11-opsdir-lc-kumari-2015-12-04-00

Request Review of draft-ietf-avtcore-multi-media-rtp-session
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Type Last Call Review
Team Ops Directorate (opsdir)
Deadline 2015-12-09
Requested 2015-11-29
Other Reviews Genart Last Call review of -11 by Meral Shirazipour (diff)
Secdir Last Call review of -11 by Christian Huitema (diff)
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Reviewer Warren Kumari
Review review-ietf-avtcore-multi-media-rtp-session-11-opsdir-lc-kumari-2015-12-04
Posted at http://www.ietf.org/mail-archive/web/ops-dir/current/msg01586.html
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Review completed: 2015-12-04

Review
review-ietf-avtcore-multi-media-rtp-session-11-opsdir-lc-kumari-2015-12-04

Be ye not afraid -- I have reviewed this document as part of the

operations directorate's ongoing effort to review all IETF documents

being processed by the IESG.  These comments were written primarily

for the benefit of the operation area directors.  Document editors and

WG chairs should treat these comments just like any other last call

comments.

Version reviewed: 

draft-ietf-avtcore-multi-media-rtp-session-11

Summary: Ready with nits. OpsAD attention not needed.

Detail:

This document allows multiple media types to be sent in a single RTP session. The changes are basically all application layer changes, and (IMO) the operations ADs can spend their time elsewhere. 

A very useful side effect of this ( / the whole point) is that there may be less transport layer flows needed, which reduces state on middleboxes like NATs and similar. This has operational implications, but as far as I can see, only good ones :-)

Nits inline below (in [O,] [P], [R] format):

--------------------------------------

AVTCORE WG                                                 M. Westerlund

Internet-Draft                                                  Ericsson

Updates: 3550, 3551 (if approved)                             C. Perkins

Intended status: Standards Track                   University of Glasgow

Expires: May 15, 2016                                          J. Lennox

Vidyo

November 12, 2015

Sending Multiple Types of Media in a Single RTP Session

draft-ietf-avtcore-multi-media-rtp-session-11

Abstract

This document specifies how an RTP session can contain RTP Streams

with media from multiple media types such as audio, video, and text.

This has been restricted by the RTP Specification, and thus this

document updates RFC 3550 and RFC 3551 to enable this behaviour for

applications that satisfy the applicability for using multiple media

types in a single RTP session.

Status of This Memo

This Internet-Draft is submitted in full conformance with the

provisions of BCP 78 and BCP 79.

Internet-Drafts are working documents of the Internet Engineering

Task Force (IETF).  Note that other groups may also distribute

working documents as Internet-Drafts.  The list of current Internet-

Drafts is at 

http://datatracker.ietf.org/drafts/current/

.

Internet-Drafts are draft documents valid for a maximum of six months

and may be updated, replaced, or obsoleted by other documents at any

time.  It is inappropriate to use Internet-Drafts as reference

material or to cite them other than as "work in progress."

This Internet-Draft will expire on May 15, 2016.

Copyright Notice

Copyright (c) 2015 IETF Trust and the persons identified as the

document authors.  All rights reserved.

This document is subject to BCP 78 and the IETF Trust's Legal

Provisions Relating to IETF Documents

(

http://trustee.ietf.org/license-info

) in effect on the date of

publication of this document.  Please review these documents

carefully, as they describe your rights and restrictions with respect

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to this document.  Code Components extracted from this document must

include Simplified BSD License text as described in Section 4.e of

the Trust Legal Provisions and are provided without warranty as

described in the Simplified BSD License.

Table of Contents

1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2

2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   3

3.  Background and Motivation . . . . . . . . . . . . . . . . . .   3

4.  Applicability . . . . . . . . . . . . . . . . . . . . . . . .   4

5.  Using Multiple Media Types in a Single RTP Session  . . . . .   6

5.1.  Allowing Multiple Media Types in an RTP Session . . . . .   6

5.2.  Demultiplexing media types within an RTP session  . . . .   7

5.3.  Per-SSRC Media Type Restrictions  . . . . . . . . . . . .   8

5.4.  RTCP Considerations . . . . . . . . . . . . . . . . . . .   9

6.  Extension Considerations  . . . . . . . . . . . . . . . . . .   9

6.1.  RTP Retransmission Payload Format . . . . . . . . . . . .   9

6.2.  RTP Payload Format for Generic FEC  . . . . . . . . . . .  10

6.3.  RTP Payload Format for Redundant Audio  . . . . . . . . .  11

7.  Signalling  . . . . . . . . . . . . . . . . . . . . . . . . .  12

8.  Security Considerations . . . . . . . . . . . . . . . . . . .  12

9.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  13

10. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  13

11. References  . . . . . . . . . . . . . . . . . . . . . . . . .  13

11.1.  Normative References . . . . . . . . . . . . . . . . . .  13

11.2.  Informative References . . . . . . . . . . . . . . . . .  14

Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  15

1.  Introduction

The Real-time Transport Protocol [RFC3550] was designed to use

separate RTP sessions to transport different types of media.  This

implies that different transport layer flows are used for different

media streams.  For example, a video conferencing application might

send audio and video traffic RTP flows on separate UDP ports.  With

increased use of network address/port translation, firewalls, and

other middleboxes it is, however, becoming difficult to establish

multiple transport layer flows between endpoints.  Hence, there is

pressure to reduce the number of concurrent transport flows used by

RTP applications.

This memo updates [RFC3550] and [RFC3551] to allow multiple media

types to be sent in a single RTP session in certain cases, thereby

reducing the number of transport layer flows that are needed.  It

makes no changes to RTP behaviour when using multiple RTP streams

containing media of the same type (e.g., multiple audio streams or

multiple video streams) in a single RTP session, however

[O] , however

[P] . However,

[R] run-on sentence

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[I-D.ietf-avtcore-rtp-multi-stream] provides important clarifications

to RTP behaviour in that case.

This memo is structured as follows.  Section 2 defines terminology.

Section 3 further describes the background to, and motivation for,

this memo and Section 4 describes the scenarios where this memo is

applicable.  Section 5 discusses issues arising from the base RTP and

RTCP specification when using multiple types of media in a single RTP

session, while Section 6 considers the impact of RTP extensions.  We

discuss signalling in Section 7.  Finally, security considerations

are discussed in Section 8.

2.  Terminology

The terms Encoded Stream, Endpoint, Media Source, RTP Session, and

RTP Stream are used as defined in

[I-D.ietf-avtext-rtp-grouping-taxonomy].  We also define the

following terms:

Media Type:  The general type of media data used by a real-time

application.  The media type corresponds to the value used in the

<media> field of an SDP m= line.  The media types defined at the

time of this writing are "audio", "video", "text", "image",

"application", and "message".  [RFC4566] [RFC6466]

Quality of Service (QoS):  Network mechanisms that are intended to

ensure that the packets within a flow or with a specific marking

are transported with certain properties.

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

[RFC2119].

3.  Background and Motivation

RTP was designed to support multimedia sessions, containing multiple

types of media sent simultaneously, by using multiple transport layer

flows.  The existence of network address translators, firewalls, and

other middleboxes complicates this, however, since a mechanism is

needed to ensure that all the transport layer flows needed by the

application can be established.  This has three consequences:

1.  increased delay to establish a complete session, since each of

the transport layer flows needs to be negotiated and established;

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2.  increased state and resource consumption in the middleboxes that

can lead to unexpected behaviour when middlebox resource limits

are reached; and

3.  increased risk that a subset of the transport layer flows will

fail to be established, thus preventing the application from

communicating.

Using fewer transport layer flows can hence be seen to reduce the

risk of communication failure, and can lead to improved reliability

and performance.

One of the benefits of using multiple transport layer flows is that

it makes it easy to use network layer quality of service (QoS)

mechanisms to give differentiated performance for different flows.

However, we note that many RTP-using application don't use network

QoS features, and don't expect or desire any separation in network

treatment of their media packets, independent of whether they are

audio, video or text.  When an application has no such desire, it

doesn't need to provide a transport flow structure that simplifies

flow based QoS.

Given the above issues, it might seem appropriate for RTP-based

applications to send all their media streams bundled into one RTP

session, running over a single transport layer flow.  However, this

is prohibited by the RTP specification, because the design of RTP

makes certain assumptions that can be incompatible with sending

multiple media types in a single RTP session.  Specifically, the RTP

control protocol (RTCP) timing rules assume that all RTP media flows

in a single RTP session have broadly similar RTCP reporting and

feedback requirements, which can be problematic when different types

of media are multiplexed together.  Various RTP extensions also make

assumptions about SSRC use and RTCP reporting that are incompatible

with sending different media types in a single RTP session.

This memo updates [RFC3550] and [RFC3551] to allow RTP sessions to

contain more than one media type in certain circumstances, and gives

guidance on when it is safe to send multiple media types in a single

RTP session.

4.  Applicability

This specification has limited applicability, and anyone intending to

use it needs to ensure that their application and use case meets the

following criteria:

Equal treatment of media:  The use of a single RTP session requires

similar network treatment for all types of media used within the

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session.  Applications that require significantly different

network quality of service (QoS) or RTCP configuration for

different media streams are better suited by sending those media

streams on separate RTP session, using separate transport layer

flows for each, since that gives greater flexibility.  Further

guidance on how to provide differential treatment for some media

is given in [I-D.ietf-avtcore-multiplex-guidelines] and

[I-D.ietf-dart-dscp-rtp].

Compatible RTCP Behaviour:  The RTCP timing rules enforce a single

RTCP reporting interval for all participants in an RTP session.

Flows with very different media sending rate or RTCP feedback

requirements cannot be multiplexed together, since this leads to

either excessive or insufficient RTCP for some flows, depending

[O]depending

[P] depending on

[R] readability

how the RTCP session bandwidth, and hence reporting interval, is

configured.  For example, it is likely not feasible to find a

single RTCP configuration that simultaneously suits both a low-

rate audio flow with no feedback and a high-quality video flow

with sophisticated RTCP-based feedback needs, making it difficult

to combine these into a single RTP session.

[O] For example, it is likely not feasible to find a

single RTCP configuration that simultaneously suits both a low-

rate audio flow with no feedback and a high-quality video flow

with sophisticated RTCP-based feedback needs, making it difficult

to combine these into a single RTP session.

[P] For example, it is likely infeasible to find a single RTCP configuration that simultaneously suits both a low-rate audio flow with no feedback, and a high-quality video flow with sophisticated RTCP-based feedback. Thus, combining these into a single RTP session is difficult and/or inadvisable.

[R] readability

Signalled Support:  The extensions defined in this memo are not

compatible with unmodified [RFC3550]-compatible endpoints.  Their

use requires signalling and mutual agreement by all participants

within an RTP session.  This requirement can be a problem for

signalling solutions that can't negotiate with all participants.

For declarative signalling solutions, mandating that the session

is using multiple media types in one RTP session can be a way of

attempting to ensure that all participants in the RTP session

follow the requirement.  However, for signalling solutions that

lack methods for enforcing that a receiver supports a specific

feature, this can still cause issues.

Consistent support for multiparty RTP sessions:  If it is desired to

send multiple types of media in a multiparty RTP session, then all

participants in that session need to support sending multiple type

of media in a single RTP session.  It is not possible, in the

general case, to implement a gateway that can interconnect an

endpoint using multiple types of media sent using separate RTP

sessions, with one or more endpoints that send multiple types of

media in a single RTP session.

One reason for this is that the same SSRC value can safely be used

for different streams in multiple RTP sessions, but when collapsed

to a single RTP session there is an SSRC collision.  This would

not be an issue, since SSRC collision detection will resolve the

conflict, except that some RTP payload formats and extensions use

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matching SSRCs to identify related flows, and break when a single

RTP session is used.

A middlebox that remaps SSRC values when combining multiple RTP

sessions into one also needs to be aware of all possible RTCP

packet types that might be used, so that it can remap the SSRC

values in those packets.  This is impossible to do without

restricting the set of RTCP packet types that can be used to those

that are known by the middlebox.  Such a middlebox might also have

difficulty due to differences in configured RTCP bandwidth and

other parameters between the RTP sessions.

Finally, the use of a middlebox that translates SSRC values can

negatively impact the possibility for loop detection, as SSRC/CSRC

can't be used to detect the loops, instead some other RTP stream

[O] loops, instead

[P] loops; instead,

[R] run on sentence

or media source identity name space that is common across all

interconnect parts are needed.

[O] are needed.

[P] is needed.

[R] grammar

Ability to operate with limited payload type space:  An RTP session

has only a single 7-bit payload type space for all its payload

type numbers.  Some applications might find this space limiting

when media different media types and RTP payload formats are using

[O] when media different media types

[P] when different media types

[R] grammar/typo

within a single RTP session.

Avoids incompatible Extensions:  Some RTP and RTCP extensions rely on

the existence of multiple RTP sessions and relate media streams

between sessions.  Others report on particular media types, and

cannot be used with other media types.  Applications that send

multiple types of media into a single RTP session need to avoid

such extensions.

5.  Using Multiple Media Types in a Single RTP Session

This section defines what needs to be done or avoided to make an RTP

session with multiple media types function without issues.

5.1.  Allowing Multiple Media Types in an RTP Session

Section 5.2 of "RTP: A Transport Protocol for Real-Time Applications"

[RFC3550] states:

For example, in a teleconference composed of audio and video media

encoded separately, each medium SHOULD be carried in a separate

RTP session with its own destination transport address.

Separate audio and video streams SHOULD NOT be carried in a single

RTP session and demultiplexed based on the payload type or SSRC

fields.

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This specification changes both of these sentences.  The first

sentence is changed to:

For example, in a teleconference composed of audio and video media

encoded separately, each medium SHOULD be carried in a separate

RTP session with its own destination transport address, unless

specification [RFCXXXX] is followed and the application meets the

applicability constraints.

The second sentence is changed to:

Separate audio and video media sources SHOULD NOT be carried in a

single RTP session, unless the guidelines specified in [RFCXXXX]

are followed.

Second paragraph of Section 6 in RTP Profile for Audio and Video

Conferences with Minimal Control [RFC3551] says:

The payload types currently defined in this profile are assigned

to exactly one of three categories or media types: audio only,

video only and those combining audio and video.  The media types

are marked in Tables 4 and 5 as "A", "V" and "AV", respectively.

Payload types of different media types SHALL NOT be interleaved or

multiplexed within a single RTP session, but multiple RTP sessions

MAY be used in parallel to send multiple media types.  An RTP

source MAY change payload types within the same media type during

a session.  See the section "Multiplexing RTP Sessions" of RFC

3550 for additional explanation.

This specifications purpose is to violate that existing SHALL NOT

[O] is to violate that existing

[P] is to specify that existing

[R] I am not sure what was intended for "violate"; maybe specify, or something else?

under certain conditions.  Thus this sentence also has to be changed

to allow for multiple media type's payload types in the same session.

The above sentence is changed to:

Payload types of different media types SHALL NOT be interleaved or

multiplexed within a single RTP session unless [RFCXXXX] is used,

and the application conforms to the applicability constraints.

Multiple RTP sessions MAY be used in parallel to send multiple

media types.

RFC-Editor Note: Please replace RFCXXXX with the RFC number of this

specification when assigned.

5.2.  Demultiplexing media types within an RTP session

When receiving packets from a transport layer flow, an endpoint will

first separate the RTP and RTCP packets from the non-RTP packets, and

pass them to the RTP/RTCP protocol handler.  The RTP and RTCP packets

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are then demultiplexed based on their SSRC into the different media

streams.  For each media stream, incoming RTCP packets are processed,

and the RTP payload type is used to select the appropriate media

decoder.  This process remains the same irrespective of whether

multiple media types are sent in a single RTP session or not.

It is important to note that the RTP payload type is never used to

distinguish media streams.  The RTP packets are demultiplexed into

media streams based on their SSRC, then the RTP payload type is used

to select the correct media decoding pathway for each media stream.

5.3.  Per-SSRC Media Type Restrictions

An SSRC in an RTP session can change between media formats of the

same type, subject to certain restrictions [RFC7160], but MUST NOT

change media type during its lifetime.  For example, an SSRC can

change between different audio formats, but cannot start sending

audio then change to sending video.  The lifetime of an SSRC ends

when an RTCP BYE packet for that SSRC is sent, or when it ceases

transmission for long enough that it times out for the other

participants in the session.

The main motivation is that a given SSRC has its own RTP timestamp

and sequence number spaces.  The same way that you can't send two

encoded streams of audio with the same SSRC, you can't send one

encoded audio and one encoded video stream with the same SSRC.  Each

encoded stream when made into an RTP 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 for that particular encoded

stream.  Nor can one easily determine which clock rate a particular

SSRCs timestamp will increase with.  For additional arguments why RTP

payload type based multiplexing of multiple media sources doesn't

work see [I-D.ietf-avtcore-multiplex-guidelines].

[O] work see

[P] work, see

[R] grammar

Within an RTP session where multiple media types have been configured

for use, an SSRC can only send one type of media during its lifetime

(i.e., it can switch between different audio codecs, since those are

both the same type of media, but cannot switch between audio and

video).  Different SSRCs MUST be used for the different media

sources, the same way multiple media sources of the same media type

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

unique across all of the payload configurations independent of media

type that is used in the RTP session.

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5.4.  RTCP Considerations

When sending multiple types of media that have different rates in a

single RTP session, endpoints MUST follow the guidelines for handling

RTCP described in Section 7 of [I-D.ietf-avtcore-rtp-multi-stream].

6.  Extension Considerations

This section outlines known issues and incompatibilities with RTP and

RTCP extensions when multiple media types are used in a single RTP

sessions.  Future extensions to RTP and RTCP need to consider, and

document, any potential incompatibility.

6.1.  RTP Retransmission Payload Format

The RTP Retransmission Payload Format [RFC4588] can operate in either

SSRC-multiplexed mode or session-multiplex mode.

In SSRC-multiplexed mode, retransmitted RTP packets are sent in the

same RTP session as the original packets, but use a different SSRC

with the same RTCP SDES CNAME.  If each endpoint sends only a single

original RTP stream and a single retransmission RTP stream in the

session, this is sufficient.  If an endpoint sends multiple original

and retransmission RTP streams, as would occur when sending multiple

media types in a single RTP session, then each original RTP stream

and the retransmission RTP stream have to be associated using

heuristics.  By having retransmission requests outstanding for only

one SSRC not yet mapped, a receiver can determine the binding between

original and retransmission RTP stream.  Another alternative is the

use of different RTP payload types, allowing the signalled "apt"

(associated payload type) parameter of the RTP retransmission payload

format to be used to associate retransmitted and original packets.

Session-multiplexed mode sends the retransmission RTP stream in a

separate RTP session to the original RTP stream, but using the same

SSRC for each, with association being done by matching SSRCs between

the two sessions.  This is unaffected by the use of multiple media

types in a single RTP session, since each media type will be sent

using a different SSRC in the original RTP session, and the same

SSRCs can be used in the retransmission session, allowing the streams

to be associated.  This can be signalled using SDP with the BUNDLE

[I-D.ietf-mmusic-sdp-bundle-negotiation] and FID grouping [RFC5888]

extensions.  These SDP extensions require each "m=" line to only be

included in a single FID group, but the RTP retransmission payload

format uses FID groups to indicate the m= lines that form an original

and retransmission pair.  Accordingly, when using the BUNDLE

extension to allow multiple media types to be sent in a single RTP

session, each original media source (m= line) that is retransmitted

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needs a corresponding m= line in the retransmission RTP session.  In

case there are multiple media lines for retransmission, these media

lines will form a independent BUNDLE group from the BUNDLE group with

the source streams.

An example SDP fragment showing the grouping structures is provided

in Figure 1.  This example is not legal SDP and only the most

important attributes have been left in place.  Note that this SDP is

not an initial BUNDLE offer.  As can be seen there are two bundle

groups, one for the source RTP session and one for the

retransmissions.  Then each of the media sources are grouped with its

retransmission flow using FID, resulting in three more groupings.

a=group:BUNDLE foo bar fiz

a=group:BUNDLE zoo kelp glo

a=group:FID foo zoo

a=group:FID bar kelp

a=group:FID fiz glo

m=audio 10000 RTP/AVP 0

a=mid:foo

a=rtpmap:0 PCMU/8000

m=video 10000 RTP/AVP 31

a=mid:bar

a=rtpmap:31 H261/90000

m=video 10000 RTP/AVP 31

a=mid:fiz

a=rtpmap:31 H261/90000

m=audio 40000 RTP/AVPF 99

a=rtpmap:99 rtx/90000

a=fmtp:99 apt=0;rtx-time=3000

a=mid:zoo

m=video 40000 RTP/AVPF 100

a=rtpmap:100 rtx/90000

a=fmtp:199 apt=31;rtx-time=3000

a=mid:kelp

m=video 40000 RTP/AVPF 100

a=rtpmap:100 rtx/90000

a=fmtp:199 apt=31;rtx-time=3000

a=mid:glo

Figure 1: SDP example of Session Multiplexed RTP Retransmission

6.2.  RTP Payload Format for Generic FEC

The RTP Payload Format for Generic Forward Error Correction (FEC)

[RFC5109] (and its predecessor [RFC2733]) can either send the FEC

stream as a separate RTP stream, or it can send the FEC combined with

the original RTP stream as a redundant encoding [RFC2198].

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When sending FEC as a separate stream, the RTP Payload Format for

generic FEC requires that FEC stream to be sent in a separate RTP

session to the original stream, using the same SSRC, with the FEC

stream being associated by matching the SSRC between sessions.  The

RTP session used for the original streams can include multiple RTP

streams, and those RTP stream can use multiple media types.  The

[O] those RTP stream

[P] those RTP streams

[R] grammar

repair session only needs one RTP Payload type to indicate FEC data,

irrespective of the number of FEC streams sent, since the SSRC is

used to associate the FEC streams with the original streams.  Hence,

it is RECOMMENDED that FEC stream use the "application/ulpfec" media

type for [RFC5109], and the "application/parityfec" media type for

[RFC2733].  It is legal, but NOT RECOMMENDED, to send FEC streams

using media specific payload format names (e.g., if an original RTP

session contains audio and video flows, for the associated FEC RTP

session where to use the "audio/ulpfec" and "video/ulpfec" payload

formats), since this unnecessarily uses up RTP payload type values,

and adds no value for demultiplexing since there might be multiple

streams of the same media type).

The combination of an original RTP session using multiple media types

with a associated generic FEC session can be signalled using SDP with

the BUNDLE extension [I-D.ietf-mmusic-sdp-bundle-negotiation].  In

this case, the RTP session carrying the FEC streams will be its own

BUNDLE group.  The m= line for each original stream and the m= line

for the corresponding FEC stream are grouped using the SDP grouping

framework using either the FEC-FR [RFC5956] grouping or, for

backwards compatibility, the FEC [RFC4756] grouping.  This is similar

to the situation that arises for RTP retransmission with session

multiplexing discussed in Section 6.1.

The Source-Specific Media Attributes [RFC5576] specification defines

an SDP extension (the "FEC" semantic of the "ssrc-group" attribute)

to signal FEC relationships between multiple RTP streams within a

single RTP session.  This cannot be used with generic FEC, since the

FEC repair packets need to have the same SSRC value as the source

packets being protected.  There is ongoing work on an ULP extension

to allow it be use FEC RTP streams within the same RTP Session as the

source stream [I-D.lennox-payload-ulp-ssrc-mux].

When the FEC is sent as a redundant encoding, the considerations in

Section 6.3 apply.

6.3.  RTP Payload Format for Redundant Audio

The RTP Payload Format for Redundant Audio [RFC2198] can be used to

protect audio streams.  It can also be used along with the generic

FEC payload format to send original and repair data in the same RTP

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packets.  Both are compatible with RTP sessions containing multiple

media types.

This payload format requires each different redundant encoding use a

different RTP payload type number.  When used with generic FEC in

sessions that contain multiple media types, this requires each media

type use a different payload type for the FEC stream.  For example,

if audio and text are sent in a single RTP session with generic ULP

FEC sent as a redundant encoding for each, then payload types need to

be assigned for FEC using the audio/ulpfec and text/ulpfec payload

formats.  If multiple original payload types of used in the session,

[O] of used

[P] are used

[R] typo?

different redundant payload types need to be allocated for each one.

This has potential to rapidly exhaust the available RTP payload type

numbers.

7.  Signalling

Establishing a single RTP session using multiple media types requires

signalling.  This signalling has to:

1.  ensure that any participant in the RTP session is aware that this

is an RTP session with multiple media types;

2.  ensure that the payload types in use in the RTP session are using

unique values, with no overlap between the media types;

3.  ensure RTP session level parameters, for example the RTCP RR and

RS bandwidth modifiers, the RTP/AVPF trr-int parameter, transport

protocol, RTCP extensions in use, and any security parameters,

are consistent across the session; and

4.  ensure that RTP and RTCP functions that can be bound to a

particular media type are reused where possible, rather than

configuring multiple code-points for the same thing.

When using SDP signalling, the BUNDLE extension

[I-D.ietf-mmusic-sdp-bundle-negotiation] is used to signal RTP

sessions containing multiple media types.

8.  Security Considerations

RTP provides a range of strong security mechanisms that can be used

to secure sessions [RFC7201], [RFC7202].  The majority of these are

independent of the type of media sent in the RTP session, however it

[O] session, however it

[P] session; however, it

[R] grammar

is important to check that the security mechanism chosen is

compatible with all types of media sent within the session.