wdiff rfc6828.original rfc6828.txt

AVTEXT Working Group
Internet Engineering Task Force (IETF) J. Xia
Internet-Draft
Request for Comments: 6828 Huawei
Intended status:
Category: Informational November 12, 2012
Expires: May 16, January 2013
ISSN: 2070-1721

Content Splicing for RTP Sessions
draft-ietf-avtext-splicing-for-rtp-12

Abstract

Content splicing is a process that replaces the content of a main
multimedia stream with other multimedia content, content and delivers the
substitutive multimedia content to the receivers for a period of
time. Splicing is commonly used for local advertisement insertion of local
advertisements by cable operators, replacing a whereby national advertisement
content is replaced with a local advertisement.

This memo describes some use cases for content splicing and a set of
requirements for splicing content delivered by RTP. It provides
concrete guidelines for how an RTP mixer can be used to handle
content splicing.

Status of this This Memo

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approved by the IESG are a maximum candidate for any level of six months Internet
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This Internet-Draft will expire on May 16, 2013.
http://www.rfc-editor.org/info/rfc6828.

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Table of Contents

1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3 ....................................................2
2. System Model and Terminology . . . . . . . . . . . . . . . . . 3 ....................................3
3. Requirements for RTP Splicing . . . . . . . . . . . . . . . . 6 ...................................6
4. Content Splicing for RTP sessions . . . . . . . . . . . . . . 7 Sessions ...............................7
4.1. RTP Processing in RTP Mixer . . . . . . . . . . . . . . . 7 ................................7
4.2. RTCP Processing in RTP Mixer . . . . . . . . . . . . . . . 8 ...............................8
4.3. Considerations for Handling Media Clipping at the
RTP Layer . . . . . . . . . . . . . . . . . . . . . . . . . . 10 .................................................10
4.4. Congestion Control Considerations . . . . . . . . . . . . 11 .........................11
4.5. Considerations for Implementing Undetectable Splicing . . 12 .....13
5. Implementation Considerations . . . . . . . . . . . . . . . . 13 ..................................13
6. Security Considerations . . . . . . . . . . . . . . . . . . . 13 ........................................14
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 14
8. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 14
9. 10. Appendix- Why Mixer Is Chosen . . . . . . . . . . . . . . 15
10. ................................................15
8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 15
10.1. .....................................................15
8.1. Normative References . . . . . . . . . . . . . . . . . . . 15
10.2. ......................................15
8.2. Informative References . . . . . . . . . . . . . . . . . . 16
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 16 ....................................15
Appendix A. Why Mixer Is Chosen ...................................17

1. Introduction

This document outlines how content splicing can be used in RTP
sessions. Splicing, in general, is a process where part of a
multimedia content is replaced with other multimedia content, content and
delivered to the receivers for a period of time. The substitutive
content can be provided provided, for example example, via another stream or via local
media file storage. One representative use case for splicing is
local advertisement insertion, allowing insertion. This allows content providers to
replace
the national advertising content with its their own regional
advertising content prior to delivering the regional advertising
content to the receivers. Besides the advertisement insertion use
case, there are other use cases in which the splicing technology can
be applied. For applied, for example, splicing a recorded video into a video
conferencing session, session or implementing a playlist server that stitches
pieces of video together.

Content splicing is a well-defined operation in MPEG-based cable TV
systems. Indeed, the Society for Cable Telecommunications Engineers
(SCTE) has created two standards, [SCTE30] and [SCTE35], to
standardize MPEG2-TS splicing procedure. procedures. SCTE 30 creates a
standardized method for communication between advertisements advertisement server
and splicer, and SCTE 35 supports splicing of MPEG2 transport
streams.

When using multimedia splicing into the internet, Internet, the media may be
transported by RTP. In this case case, the original media content and
substitutive media content will use the same time period, period but may
contain different numbers of RTP packets due to different media
codecs and entropy coding. This mismatch may require some
adjustments of the RTP header sequence number to maintain
consistency. [RFC3550] provides the tools to enabled enable seamless content
splicing in RTP session, sessions, but to date there has have been no clear
guidelines on how to use these tools.

This memo outlines the requirements for content splicing in RTP
sessions and describes how an RTP mixer can be used to meet these
requirements.

2. System Model and Terminology

In this document, the splicer, an intermediary network element, the Splicer
handles RTP splicing. The Splicer splicer can receive main content and
substitutive content simultaneously, simultaneously but will send one of them at one
point of time.

When RTP splicing begins, the splicer sends the substitutive content
to the RTP receiver instead of the main content for a period of time.
When RTP splicing ends, the splicer switches back to sending the main
content to the RTP receiver.

A simplified RTP splicing diagram is depicted in Figure 1, in which
only one main content flow and one substitutive content flow are
given. Actually, the splicer can handle multiple splicing for
multiple RTP sessions simultaneously. RTP splicing may happen more
than once in multiple time slots during the lifetime of the main RTP
stream. The methods how by which the splicer learns when to start and
end the splicing is are out of scope for this document.

+---------------+
| | Main Content +-----------+
| Main RTP |------------->| | Output Content
| Content | | Splicer |--------------->
+---------------+ ---------->| |
| +-----------+
|
| Substitutive Content
|
|
+-----------------------+
| Substitutive RTP |
| Content |
| or |
| Local File Storage |
+-----------------------+

Figure 1: RTP Splicing Architecture

This document uses the following terminologies.

Output RTP Stream

The RTP stream that the RTP receiver is currently receiving. The
content of the output of the RTP stream can be either main content
or substitutive content.

Main Content

The multimedia content that are is conveyed in the main RTP stream.
Main content will be replaced by the substitutive content during
splicing.

Main RTP Stream

The RTP stream that the splicer is receiving. The content of the
main RTP stream can be replaced by substitutive content for a
period of time.

Main RTP Sender

The sender of RTP packets carrying the main RTP stream.

Substitutive Content

The multimedia content that replaces the main content during
splicing. The substitutive content can can, for example example, be contained
in an RTP stream from a media sender or fetched from local media
file storage.

Substitutive RTP Stream

An RTP stream with new content that will replace the content in
the main RTP stream. Substitutive The substitutive RTP stream and main RTP
stream are two separate streams. If the substitutive content is
provided via a substitutive RTP stream, the substitutive RTP Stream
stream must pass through the splicer before the substitutive
content is delivered to the receiver.

Substitutive RTP Sender

The sender of RTP packets carrying the substitutive RTP stream.

Splicing In

Splicing-In Point

A virtual point in the RTP stream, suitable for substitutive
content entry, typically in the boundary between two independently
decodable frames.

Splicing Out

Splicing-Out Point

A virtual point in the RTP stream, suitable for substitutive
content exist, exit, typically in the boundary between two independently
decodable frames.

Splicer

An intermediary node that inserts substitutive content into a main
RTP stream. The splicer sends substitutive content to the RTP
receiver instead of main content during splicing. It is also
responsible for processing RTCP RTP Control Protocol (RTCP) traffic
between the RTP sender and the RTP receiver.

3. Requirements for RTP Splicing

In order to allow seamless content splicing at the RTP layer, the
following requirements must be met. Meeting these will also allow,
but not require, seamless content splicing at layers above RTP.

REQ-1:

The splicer should be agnostic about the network and transport
layer
transport-layer protocols used to deliver the RTP streams.

REQ-2:

The splicing operation at the RTP layer must allow splicing at any
point required by the media content, content and must not constrain when
splicing in
splicing-in or splicing out splicing-out operations can take place.

REQ-3:

Splicing of RTP content must be backward compatible with the RTP/
RTCP
RTP/RTCP protocol, associated profiles, payload formats, and
extensions.

REQ-4:

The splicer will modify the content of RTP packets, packets and thus break
the end-to-end security, at a minimum minimum, breaking the data integrity
and source authentication. If the Splicer splicer is designated to insert
substitutive content, it must be trusted, i.e., be in the security
context(s) with the main RTP sender, the substitutive RTP sender,
and the receivers. If encryption is employed, the splicer
commonly must decrypt the inbound RTP packets and re-encrypt the
outbound RTP packets after splicing.

REQ-5:

The splicer should rewrite as necessary and forward RTCP messages
(e.g., including packet loss, jitter, etc.) sent from a downstream
receiver to the main RTP sender or the substitutive RTP sender,
and thus allow the main RTP sender or substitutive RTP sender to
learn the performance of the downstream receiver when its content
is being passed to an RTP receiver. In addition, the splicer
should rewrite RTCP messages from the main RTP sender or
substitutive RTP sender to the receiver.

REQ-6:

The splicer must not affect other RTP sessions running between the
RTP sender and the RTP receiver, receiver and must be transparent for the
RTP sessions it does not splice.

REQ-7:

The RTP receiver should not be able to detect any splicing points
in the RTP stream produced by the splicer on the RTP protocol
level. For the advertisement insertion use case, it is important
to make it difficult for the RTP receiver to detect where an
advertisement insertion is starting or ending from the RTP
packets, and thus avoiding the RTP receiver from filtering out the
advertisement content. This memo only focuses on making the
splicing undetectable at the RTP layer. The corresponding
processing is depicted in section Section 4.5.

4. Content Splicing for RTP sessions Sessions

The RTP specification [RFC3550] defines two types of middlebox: middleboxes: RTP
translators and RTP mixers. Splicing is best viewed as a mixing
operation. The splicer generates a new RTP stream that is a mix of
the main RTP stream and the substitutive RTP stream. An RTP mixer is
therefore an appropriate model for a content splicer. In the next
four subsections (from subsection Section 4.1 to subsection Section 4.4), the document
analyzes how the mixer handles RTP splicing and how it satisfies the
general requirements listed in section Section 3. In subsection Section 4.5, the
document looks at REQ-7 in order to hide the fact that splicing take takes
place.

4.1. RTP Processing in RTP Mixer

A splicer could be implemented as a mixer that receives the main RTP
stream and the substitutive content (possibly via a substitutive RTP
stream), and sends a single output RTP stream to the receiver(s).
That output RTP stream will contain either the main content or the
substitutive content. The output RTP stream will come from the
mixer, mixer
and will have the synchronization source (SSRC) of the mixer rather
than the main RTP sender or the substitutive RTP sender.

The mixer uses its own SSRC, sequence number space space, and timing model
when generating the output stream. Moreover, the mixer may insert
the SSRC of the main RTP stream into the contributing source (CSRC)
list in the output media stream.

At the splicing in splicing-in point, when the substitutive content becomes
active, the mixer chooses the substitutive RTP stream as the input
stream
at splicing in point, and extracts the payload data (i.e., substitutive content).
If the substitutive content comes from local media file storage, the
mixer directly fetches the substitutive content. After that, the
mixer encapsulates substitutive content instead of main content as
the payload of the output media stream, stream and then sends the output RTP
media stream to the receiver. The mixer may insert the SSRC of the
substitutive RTP stream into the CSRC list in the output media
stream. If the substitutive content comes from local media file
storage, the mixer should leave the CSRC list blank.

At the splicing out splicing-out point, when the substitutive content ends, the
mixer retrieves the main RTP stream as the input stream at splicing out
point, and extracts
the payload data (i.e., main content). After that, the mixer
encapsulates main content instead of substitutive content as the
payload of the output media stream, stream and then sends the output media
stream to the receivers. Moreover, the mixer may insert the SSRC of
the main RTP stream into the CSRC list in the output media stream as
before.

Note that if the content is too large to fit into RTP packets sent to
the RTP receiver, the mixer needs to transcode or perform application-
layer
application-layer fragmentation. Usually the mixer is deployed as
part of a managed system and MTU will be carefully managed by this
system. This document does not raise any new MTU related issues
compared to a standard mixer described in [RFC3550].

Splicing may occur more than once during the lifetime of the main RTP
stream, this
stream. This means the mixer needs to send main content and
substitutive content in turn with its own SSRC identifier. From
receiver point of view, the only source of the output stream is the
mixer regardless of where the content is coming from.

4.2. RTCP Processing in RTP Mixer

By monitoring available bandwidth and buffer levels and by computing
network metrics such as packet loss, network jitter, and delay, an
RTP receiver can learn the network performance and communicate this
to the RTP sender via RTCP reception reports.

According to the description in section Section 7.3 of [RFC3550], the mixer
splits the RTCP flow between the sender and receiver into two
separate RTCP loops, loops; the RTP sender has no idea about the situation
on the receiver. But splicing is a processing that process where the mixer selects
one media stream from multiple streams rather than mixing them, so
the mixer can leave the SSRC identifier in the RTCP report intact
(i.e., the SSRC of the downstream receiver), this receiver). This enables the main
RTP sender or the substitutive RTP sender to learn the situation on
the receiver.

If the RTCP report corresponds to a time interval that is entirely
main content or entirely substitutive content, the number of output
RTP packets containing substitutive content is equal to the number of
input substitutive RTP packets (from the substitutive RTP stream)
during
splicing, in splicing. In the same manner, the number of output RTP
packets containing main content is equal to the number of input main
RTP packets (from the main RTP stream) during non-splicing unless the
mixer
fragment fragments the input RTP packets. This means that the mixer
does not need to modify the loss packet fields in reception report
blocks in RTCP reports. But But, if the mixer fragments the input RTP
packets, it may need to modify the loss packet fields to compensate
for the fragmentation. Whether the input RTP packets are fragmented
or not, the mixer still needs to change the SSRC field in the report
block to the SSRC identifier of the main RTP sender or the
substitutive RTP
sender, sender and rewrite the extended highest sequence
number field to the corresponding original extended highest sequence
number before forwarding the RTCP report to the main RTP sender or
the substitutive RTP sender.

If the RTCP report spans the splicing in splicing-in point or the splicing out splicing-out
point, it reflects the characteristics of the combination of main RTP
packets and substitutive RTP packets. In this case, the mixer needs
to divide the RTCP report into two separate RTCP reports and send
them to their original RTP senders senders, respectively. For each RTCP
report, the mixer also needs to make the corresponding changes to the
packet loss fields in the report block besides the SSRC field and the
extended highest sequence number field.

If the mixer receives an RTCP extended report (XR) block, it should
rewrite the XR report block in a similar way to the reception report
block in the RTCP report.

Besides forwarding the RTCP reports sent from the RTP receiver, the
mixer can also generate its own RTCP reports to inform the main RTP sender
sender, or the substitutive RTP sender sender, of the reception quality of the
content reaches the mixer when the content is not sent to the RTP
receiver. receiver when it reaches the mixer.
These RTCP reports use the SSRC of the mixer. If the substitutive
content comes from local media file storage, the mixer does not need
to generate RTCP reports for the substitutive stream.

Based on the above RTCP operating mechanism, the RTP sender whose
content is being passed to a receiver will see the reception quality
of its stream as received by the mixer, mixer and the reception quality of
the spliced stream as received by the receiver. The RTP sender whose
content is not being passed to a receiver will only see the reception
quality of its stream as received by the mixer.

The mixer must forward RTCP SDES source description (SDES) and BYE packets
from the receiver to the sender, sender and may forward them in inverse
direction as defined in
section Section 7.3 of [RFC3550].

Once the mixer receives an RTP/AVPF RTP/Audio-Visual Profile with Feedback
(AVPF) [RFC4585] transport layer transport-layer feedback packet, it must handle it carefully
carefully, as the feedback packet may contain the information of the
content that come comes from different RTP senders. In this case case, the
mixer needs to divide the feedback packet into two separate feedback
packets and process the information in the feedback control
information (FCI) in the two feedback packets, just as in the RTCP
report process described above.

If the substitutive content comes from local media file storage
(i.e., the mixer can be regarded as the substitutive RTP sender), any
RTCP packets received from downstream relate related to the substitutive
content must be terminated on the mixer without any further
processing.

4.3. Considerations for Handling Media Clipping at the RTP Layer

This section provides informative guidelines on how to handle media
substitution at both the RTP layer to minimize media impact. Dealing well
with the media substitution well at the RTP layer is necessary for quality
implementations. To perfectly erase any media impact needs more
considerations at the higher layers, how layers. How the media substitution is
erased at the higher layers are is outside of the scope of this memo.

If the time duration for any substitutive content mismatches, i.e.,
shorter or longer, longer than the duration of the main content to be
replaced, then media degradations may occur at the splicing point and
thus impact the user's experience.

If the substitutive content has shorter duration from the main
content, then there could be a gap in the output RTP stream. The RTP
sequence number will be contiguous across this gap, but there will be
an unexpected jump in the RTP timestamp. Such a gap would cause the
receiver to have nothing to play. This may be unavoidable, unless
the mixer can adjusts the splice in or splice out point to
compensate. This assumes the splicing mixer can send more of the
main RTP stream in place of the shorter substitutive stream, stream or vary
the length of the substitutive content. It is the responsibility of
the higher layer higher-layer protocols and the media providers to ensure that the
substitutive content is of very similar duration as the main content
to be replaced.

If the substitute content has longer duration than the reserved gap
duration, there will be an overlap between the substitutive RTP
stream and the main RTP stream at the splicing out splicing-out point. A
straightforward approach is that the mixer performs an ungraceful
action, terminating
action and terminates the splicing and switching switches back to the main RTP
stream even if this may cause media stuttering on the receiver.
Alternatively, the mixer may transcode the substitutive content to
play at a faster rate than normal, to adjust it to the length of the
gap in the main content, content and generate a new RTP stream for the
transcoded content. This is a complex operation, operation and very specific to
the content and media codec used. Additional approaches exists, exist; these
types of issues should be taken into account in both mixer
implementors and media generators to enable smooth substitutions.

4.4. Congestion Control Considerations

If the substitutive content has somewhat different characteristics
from the main content it replaces, or if the substitutive content is
encoded with a different codec or has different encoding bitrate, it
might overload the network and might cause network congestion on the
path between the mixer and the RTP receiver(s) that would not have
been caused by the main content.

To be robust to network congestion and packet loss, a mixer that is
performing splicing must continuously monitor the status of a
downstream network by monitoring any of the following RTCP reports
that are used:

1. RTCP receiver reports indicate packet loss [RFC3550].

2. RTCP NACKs for lost packet recovery [RFC4585].

3. RTCP ECN Explicit Congestion Notification (ECN) Feedback information
[RFC6679].

Once the mixer detects congestion on its downstream link, it will
treat these reports as follows:

1. If the mixer receives the RTCP receiver reports with packet loss
indication, it will forward the reports to the substitutive RTP
sender or the main RTP sender as described in section Section 4.2.

2. If mixer receives the RTCP NACK packets defined in [RFC4585] from
the RTP receiver for packet loss recovery, it first identifies
the content category of lost packets to which the NACK
corresponds. Then, the mixer will generate new RTCP NACK NACKs for
the lost packets with its own SSRC, SSRC and make corresponding changes
to their sequence numbers to match original, pre-spliced,
packets. If the lost substitutive content comes from local media
file storage, the mixer acting as the substitutive RTP sender
will directly fetch the lost substitutive content and retransmit
it to the RTP receiver. The mixer may buffer the sent RTP
packets and do the retransmission.

It is somewhat complex that the lost packets requested in a
single RTCP NACK message not only contain the main content but
also the substitutive content. To address this, the mixer must
divide the RTCP NACK packet into two separate RTCP NACK packets:
one requests for the lost main content, and another requests for
the lost substitutive content.

3. If an ECN-aware mixer receives RTCP ECN feedbacks feedback (RTCP ECN
feedback packets or RTCP XR summary reports) defined in [RFC6679]
from the RTP receiver, it must process them in a similar way to
the RTP/AVPF feedback packet or RTCP XR process described in
section
Section 4.2 of this memo.

These three methods require the mixer to run a congestion control
loop and bitrate adaptation between itself and the RTP receiver. The
mixer can thin or transcode the main RTP stream or the substitutive
RTP stream, but such operations are very inefficient and difficult,
and they also bring undesirable delay. Fortunately Fortunately, as noted in this
memo, the mixer acting as a splicer can rewrite the RTCP packets sent
from the RTP receiver and forward them to the RTP sender, thus
letting the RTP sender knows that congestion is being experienced on
the path between the mixer and the RTP receiver. Then, the RTP
sender applies its congestion control algorithm and reduces the media
bitrate to a value that is in compliance with congestion control
principles for the slowest link. The congestion control algorithm
may be a TCP-friendly bitrate adaptation algorithm specified in [RFC5348],
[RFC5348] or a DCCP Datagram Congestion Control Protocol (DCCP) congestion
control algorithms algorithm defined in [RFC5762].

If the substitutive content comes from local media file storage, the
mixer must directly reduce the bitrate as if it were the substitutive
RTP sender.

From the above analysis, to reduce the risk of congestion and remain
maintain the bandwidth consumption stable over time, the substitutive
RTP stream is recommended to be encoded at an appropriate bitrate to
match that of the main RTP stream. If the substitutive RTP stream
comes from the substitutive RTP sender, this sender had better has should have some
knowledge about the media encoding bitrate of the main content in
advance. How it
knows that Acquiring such knowledge is out of scope in this draft. document.

4.5. Considerations for Implementing Undetectable Splicing

If it is desirable to prevent receivers from detecting that splicing
is occurring at the RTP layer, the mixer must not include a CSRC list
in outgoing RTP packets, packets and must not forward RTCP messages from the
main RTP sender or from the substitutive RTP sender. Due to the
absence of a CSRC list in the output RTP stream, the RTP receiver
only initiates SDES, BYE BYE, and APP Application-specific functions (APP)
packets to the mixer without any knowledge of the main RTP sender and
the substitutive RTP sender.

The CSRC list identifies the contributing sources, sources; these SSRC
identifiers of contributing sources are kept globally unique for each
RTP session. The uniqueness of the SSRC identifier is used to
resolve collisions and detecting to detect RTP-level forwarding loops as
defined in
section Section 8.2 of [RFC3550]. The absence of CSRC list in this case will
create a A danger that loops involving
those contributing sources could will not be detected. detected will be created by
the absence of a CSRC list in this case. The loops could occur if
either the mixer is misconfigured to form a loop, loop or a second
mixer/translator is added, causing packets to loop back to upstream
of the original mixer. An undetected RTP packet loop is a serious denial of service
denial-of-service threat, which can consume all available bandwidth
or mixer processing resources until the looped packets are dropped as
a result of congestion. So Non-RTP So, non-RTP means must be used to detect and
resolve loops if the mixer does not add a CSRC list.

5. Implementation Considerations

When the mixer is used to handle RTP splicing, the RTP receiver does
not need any RTP/RTCP extension for splicing. As a trade-off,
additional overhead could be induced on the mixer mixer, which uses its own
sequence number space and timing model. So the mixer will rewrite
the RTP sequence number and timestamp timestamp, whatever splicing is active or
not, and generate RTCP flows for both sides. In case the mixer
serves multiple main RTP streams simultaneously, this may lead to
more overhead on the mixer.

If an undetectable splicing requirement is required, the CSRC list is
not included in the outgoing RTP packet, packet; this brings a potential
issue with loop detection as briefly described in section Section 4.5.

6. Security Considerations

The splicing application is subject to the general security
considerations of the RTP specification [RFC3550].

The mixer acting as splicer replaces some content with other content
in RTP packets, thus breaking any RTP level RTP-level end-to-end security, such
as integrity protection and source authentication. Thus Thus, any RTP
level
RTP-level or outside security mechanism, such as IPSec IPsec [RFC4301] or DTLS
Datagram Transport Layer Security [RFC6347], will use a security
association between the splicer and the receiver. When using SRTP the
Secure Real-Time Transport Protocol (SRTP) [RFC3711], the splicer
could be provisioned with the same security association as the main
RTP sender. Using a limitation in the SRTP security services
regarding source authentication, the splicer can modify and
re-protect the RTP packets without enabling the receiver to detect if
the data comes from the original source or from the splicer.

Security goals to have source authentication all the way from the RTP
main sender to the receiver through the splicer is not possible with
splicing and any existing solutions. A new solution can
theoretically be developed that enables identifying the participating
entities and what each provides, i.e. i.e., the different media sources,
main and substituting, and the splicer providing the RTP level RTP-level
integration of the media payloads in a common timeline and
synchronization context. Such a solution would obviously not meet
Req-7
REQ-7 and will be detectable on the RTP level.

The nature of this RTP service offered by a network operator
employing a content splicer is that the RTP layer RTP-layer security
relationship is between the receiver and the splicer, and between the
senders
sender and the splicer, are but is not end-to-end. end-to-end between the receiver
and the sender. This appears to invalidate the undetectability goal,
but in the common case case, the receiver will consider the splicer as the
main media source.

Some RTP deployments use RTP payload security mechanisms (e.g.,
ISMACryp [ISMACryp]). If any payload internal security mechanisms
are used, only the RTP sender and the RTP receiver establish that
security context, in which case, case any middlebox (e.g., splicer) between
the RTP sender and the RTP receiver will not get such keying
material. This may impact the splicer's possibility ability to perform splicing
if it is dependent on RTP payload level payload-level hints for finding the splice
in and out points. However, other potential solutions exist to
specify or mark where the splicing points exist in the media streams.
When using RTP payload security mechanisms mechanisms, SRTP or other security mechanism
mechanisms at RTP or lower layers can be used to provide integrity
and source authentication between the splicer and the RTP receiver.

7. IANA Considerations

No IANA actions are required.

8. Acknowledgments

The following individuals have reviewed the earlier versions of this
specification and provided very valuable comments: Colin Perkins,
Magnus Westerlund, Roni Even, Tom Van Caenegem, Joerg Ott, David R R.
Oran, Cullen Jennings, Ali C C. Begen, Charles Eckel Eckel, and Ning Zong.

9. 10. Appendix- Why Mixer Is Chosen

Translator

8. References

8.1. Normative References

[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.

[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J.
Rey, "Extended RTP Profile for Real-time Transport
Control Protocol (RTCP)-Based Feedback (RTP/AVPF)",
RFC 4585, July 2006.

[RFC6679] Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P.,
and K. Carlberg, "Explicit Congestion Notification (ECN)
for RTP over UDP", RFC 6679, August 2012.

8.2. Informative References

[ISMACryp] Internet Streaming Media Alliance (ISMA), "ISMA
Encryption and Authentication Specification 2.0",
November 2007.

[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol
(SRTP)", RFC 3711, March 2004.

[RFC4301] Kent, S. and K. Seo, "Security Architecture for the
Internet Protocol", RFC 4301, December 2005.

[RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification",
RFC 5348, September 2008.

[RFC5762] Perkins, C., "RTP and the Datagram Congestion Control
Protocol (DCCP)", RFC 5762, April 2010.

[RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security Version 1.2", RFC 6347, January 2012.

[SCTE30] Society of Cable Telecommunications Engineers (SCTE),
"Digital Program Insertion Splicing API", 2009.

[SCTE35] Society of Cable Telecommunications Engineers (SCTE),
"Digital Program Insertion Cueing Message for Cable",
2011.

Appendix A. Why Mixer Is Chosen

Both a translator and mixer both can realize splicing by changing a set of
RTP parameters.

Translator

A translator has no SSRC, SSRC; hence it is transparent to the RTP sender
and receiver. Therefore, the RTP sender sees the full path to the
receiver when the translator is passing its content. When a
translator insert inserts the substitutive content content, the RTP sender could get
a report on the path up to the translator itself. Additionally, if
splicing does not occur yet, the translator does not need to rewrite
the RTP header, and the overhead on the translator can be avoided.

If a mixer is used to do splicing, it can also allow the RTP sender
to learn the situation of its content on the receiver or on the mixer
just like the translator does, which is specified in section Section 4.2.
Compared to the translator, the mixer's outstanding benefit is that
it is pretty straight
forward straightforward to do with RTCP messages, for example,
bit-rate adaptation to handle varying network conditions. But the
translator needs more
considerations considerations, and its implementation is more
complex.

From the above analysis, both the translator and mixer have their own
advantages: less overhead or less complexity on handling RTCP.
Through After
long and sophisticated discussion, discussions, the avtext WG members decided
that they prefer less complexity rather than less overhead and incline are
inclined to choose a mixer to do splicing.

If one chooses a mixer as splicer, the overhead on the mixer must be
taken into account even if the splicing does has not occur occurred yet.

10. References

10.1. Normative References

[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.

[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
July 2006.

[RFC6679] Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P.,
and K. Carlberg, "Explicit Congestion Notification (ECN)
for RTP over UDP", RFC 6679, August 2012.

10.2. Informative References

[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.

[RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification",
RFC 5348, September 2008.

[RFC5762] Perkins, C., "RTP and the Datagram Congestion Control
Protocol (DCCP)", RFC 5762, April 2010.

[SCTE30] Society of Cable Telecommunications Engineers (SCTE),
"Digital Program Insertion Splicing API", 2009.

[SCTE35] Society of Cable Telecommunications Engineers (SCTE),
"Digital Program Insertion Cueing Message for Cable",
2011.

[ISMACryp]
Internet Streaming Media Alliance (ISMA), "ISMA Encryption
and Authentication Specification 2.0", November 2007.

Author's Address

Jinwei Xia
Huawei
Software No.101
Nanjing, Yuhuatai District 210012
China

Phone: +86-025-86622310
Email:
EMail: xiajinwei@huawei.com