Content Splicing for RTP Sessions
RFC 6828
| Document | Type | RFC - Informational (January 2013; No errata) | |
|---|---|---|---|
| Author | Jinwei Xia | ||
| Last updated | 2018-12-20 | ||
| Replaces | draft-xia-avtext-splicing-for-rtp | ||
| Stream | Internet Engineering Task Force (IETF) | ||
| Formats | plain text html pdf htmlized bibtex | ||
| Stream | WG state | Submitted to IESG for Publication | |
| Document shepherd | Magnus Westerlund | ||
| Shepherd write-up | Show (last changed 2012-10-01) | ||
| IESG | IESG state | RFC 6828 (Informational) | |
| Action Holders |
(None)
|
||
| Consensus Boilerplate | Unknown | ||
| Telechat date | |||
| Responsible AD | Gonzalo Camarillo | ||
| IESG note | Magnus Westerlund (magnus.westerlund@ericsson.com) is the document shepherd. | ||
| Send notices to | (None) | ||
Internet Engineering Task Force (IETF) J. Xia
Request for Comments: 6828 Huawei
Category: Informational January 2013
ISSN: 2070-1721
Content Splicing for RTP Sessions
Abstract
Content splicing is a process that replaces the content of a main
multimedia stream with other multimedia content and delivers the
substitutive multimedia content to the receivers for a period of
time. Splicing is commonly used for insertion of local
advertisements by cable operators, 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 Memo
This document is not an Internet Standards Track specification; it is
published for informational purposes.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Not all documents
approved by the IESG are a candidate for any level of Internet
Standard; see Section 2 of RFC 5741.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/rfc6828.
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Copyright Notice
Copyright (c) 2013 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
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. System Model and Terminology ....................................3
3. Requirements for RTP Splicing ...................................6
4. Content Splicing for RTP Sessions ...............................7
4.1. RTP Processing in RTP Mixer ................................7
4.2. RTCP Processing in RTP Mixer ...............................8
4.3. Considerations for Handling Media Clipping at the
RTP Layer .................................................10
4.4. Congestion Control Considerations .........................11
4.5. Considerations for Implementing Undetectable Splicing .....13
5. Implementation Considerations ..................................13
6. Security Considerations ........................................14
7. Acknowledgments ................................................15
8. References .....................................................15
8.1. Normative References ......................................15
8.2. Informative References ....................................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 and
delivered to the receivers for a period of time. The substitutive
content can be provided, for example, via another stream or via local
media file storage. One representative use case for splicing is
local advertisement insertion. This allows content providers to
replace national advertising content with 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
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be applied, for example, splicing a recorded video into a video
conferencing 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 procedures. SCTE 30 creates a
standardized method for communication between advertisement server
and splicer, and SCTE 35 supports splicing of MPEG2 transport
streams.
When using multimedia splicing into the Internet, the media may be
transported by RTP. In this case, the original media content and
substitutive media content will use the same time 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 enable seamless content
splicing in RTP sessions, but to date there 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,
handles RTP splicing. The splicer can receive main content and
substitutive content 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 by which the splicer learns when to start and
end the splicing are out of scope for this document.
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+---------------+
| | 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 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.
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Substitutive Content
The multimedia content that replaces the main content during
splicing. The substitutive content can, for 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. 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 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 Point
A virtual point in the RTP stream, suitable for substitutive
content entry, typically in the boundary between two independently
decodable frames.
Splicing-Out Point
A virtual point in the RTP stream, suitable for substitutive
content 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 RTP Control Protocol (RTCP) traffic
between the RTP sender and the RTP receiver.
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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 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 and must not constrain when
splicing-in or splicing-out operations can take place.
REQ-3:
Splicing of RTP content must be backward compatible with the
RTP/RTCP protocol, associated profiles, payload formats, and
extensions.
REQ-4:
The splicer will modify the content of RTP packets and thus break
the end-to-end security, at a minimum, breaking the data integrity
and source authentication. If the 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.
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REQ-6:
The splicer must not affect other RTP sessions running between the
RTP sender and the RTP 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 4.5.
4. Content Splicing for RTP Sessions
The RTP specification [RFC3550] defines two types of 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 Section 4.1 to Section 4.4), the document
analyzes how the mixer handles RTP splicing and how it satisfies the
general requirements listed in Section 3. In Section 4.5, the
document looks at REQ-7 in order to hide the fact that splicing 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
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, 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.
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At the splicing-in point, when the substitutive content becomes
active, the mixer chooses the substitutive RTP stream as the input
stream 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 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 point, when the substitutive content ends, the
mixer retrieves the main RTP stream as the input stream 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 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 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 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 7.3 of [RFC3550], the mixer
splits the RTCP flow between the sender and receiver into two
separate RTCP loops; the RTP sender has no idea about the situation
on the receiver. But splicing is a 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
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(i.e., the SSRC of the downstream 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 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 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, 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 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 point or the 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, 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, or the substitutive RTP sender, of the reception quality of
content not sent to the RTP 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.
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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 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 source description (SDES) and BYE packets
from the receiver to the sender and may forward them in inverse
direction as defined in Section 7.3 of [RFC3550].
Once the mixer receives an RTP/Audio-Visual Profile with Feedback
(AVPF) [RFC4585] transport-layer feedback packet, it must handle it
carefully, as the feedback packet may contain the information of the
content that comes from different RTP senders. In this 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 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 the RTP layer to minimize media impact. Dealing well
with the media substitution at the RTP layer is necessary for quality
implementations. To perfectly erase any media impact needs more
considerations at the higher layers. How the media substitution is
erased at the higher layers is outside of the scope of this memo.
If the time duration for any substitutive content mismatches, i.e.,
shorter or 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 or vary
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the length of the substitutive content. It is the responsibility of
the 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 point. A
straightforward approach is that the mixer performs an ungraceful
action and terminates the splicing and 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 and generate a new RTP stream for the
transcoded content. This is a complex operation and very specific to
the content and media codec used. Additional approaches 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 Explicit Congestion Notification (ECN) Feedback information
[RFC6679].
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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 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 NACKs for
the lost packets with its own 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 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 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, 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] or a Datagram Congestion Control Protocol (DCCP) congestion
control algorithm defined in [RFC5762].
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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
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 should have some
knowledge about the media encoding bitrate of the main content in
advance. Acquiring such knowledge is out of scope in this 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 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, and 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; 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 to detect RTP-level forwarding loops as
defined in Section 8.2 of [RFC3550]. A danger that loops involving
those contributing sources will not be 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 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 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 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, which uses its own
sequence number space and timing model. So the mixer will rewrite
the RTP sequence number and 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.
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If an undetectable splicing requirement is required, the CSRC list is
not included in the outgoing RTP packet; this brings a potential
issue with loop detection as briefly described in 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 end-to-end security, such
as integrity protection and source authentication. Thus, any
RTP-level or outside security mechanism, such as IPsec [RFC4301] or
Datagram Transport Layer Security [RFC6347], will use a security
association between the splicer and the receiver. When using 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., the different media sources,
main and substituting, and the splicer providing the RTP-level
integration of the media payloads in a common timeline and
synchronization context. Such a solution would obviously not meet
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 security
relationship is between the receiver and the splicer, and between the
sender and the splicer, but is not end-to-end between the receiver
and the sender. This appears to invalidate the undetectability goal,
but in the common 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 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 ability to perform splicing
if it is dependent on RTP payload-level hints for finding the splice
in and out points. However, other potential solutions exist to
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specify or mark where the splicing points exist in the media streams.
When using RTP payload security mechanisms, SRTP or other security
mechanisms at RTP or lower layers can be used to provide integrity
and source authentication between the splicer and the RTP receiver.
7. 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.
Oran, Cullen Jennings, Ali C. Begen, Charles Eckel, and Ning Zong.
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.
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RFC 6828 RTP Splicing January 2013
[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.
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RFC 6828 RTP Splicing January 2013
Appendix A. Why Mixer Is Chosen
Both a translator and mixer can realize splicing by changing a set of
RTP parameters.
A translator has no 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 inserts the substitutive 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 4.2.
Compared to the translator, the mixer's outstanding benefit is that
it is pretty straightforward to do with RTCP messages, for example,
bit-rate adaptation to handle varying network conditions. But the
translator needs more 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. After
long and sophisticated discussions, the avtext WG members decided
that they prefer less complexity rather than less overhead and 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 has not occurred yet.
Author's Address
Jinwei Xia
Huawei
Software No.101
Nanjing, Yuhuatai District 210012
China
Phone: +86-025-86622310
EMail: xiajinwei@huawei.com
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