Internet DRAFT - draft-ietf-avtext-splicing-for-rtp
draft-ietf-avtext-splicing-for-rtp
AVTEXT Working Group J. Xia
Internet-Draft Huawei
Intended status: Informational November 14, 2012
Expires: May 18, 2013
Content Splicing for RTP Sessions
draft-ietf-avtext-splicing-for-rtp-13
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 local advertisement insertion by
cable operators, replacing a national advertisement content 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
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This Internet-Draft will expire on May 18, 2013.
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document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
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 . . 12
5. Implementation Considerations . . . . . . . . . . . . . . . . 13
6. Security Considerations . . . . . . . . . . . . . . . . . . . 13
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 14
8. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 14
9. 10. Appendix- Why Mixer Is Chosen . . . . . . . . . . . . . . 15
10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 15
10.1. Normative References . . . . . . . . . . . . . . . . . . . 15
10.2. Informative References . . . . . . . . . . . . . . . . . . 16
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 16
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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, allowing content providers to replace
the national advertising content with its 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 splicing technology can 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 procedure. SCTE 30 creates a
standardized method for communication between advertisements 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 enabled seamless
content splicing in RTP session, but to date there has 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, an intermediary network element, the Splicer
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
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to the RTP receiver instead of the main content for a period of time.
When RTP splicing ends, the splicer switches back 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 splicer learns when to start and end the
splicing is 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 output RTP stream can be either main content or
substitutive content.
Main Content
The multimedia content that are conveyed in main RTP stream. Main
content will be replaced by the substitutive content during
splicing.
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Main RTP Stream
The RTP stream that the splicer is receiving. The content of 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 for example be contained
in an RTP stream from a media sender or fetched from local media
file storage.
Substitutive RTP Stream
A RTP stream with new content that will replace the content in the
main RTP stream. Substitutive RTP stream and main RTP stream are
two separate streams. If the substitutive content is provided via
substitutive RTP stream, the substitutive RTP Stream must pass
through the splicer before the substitutive content is delivered
to 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 exist, typically in the boundary between two independently
decodable frames.
Splicer
An intermediary node that inserts substitutive content into main
RTP stream. The splicer sends substitutive content to RTP
receiver instead of main content during splicing. It is also
responsible for processing RTCP traffic between the RTP sender and
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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 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 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 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 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 middlebox: 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 next four
subsections (from subsection 4.1 to subsection 4.4), the document
analyzes how the mixer handles RTP splicing and how it satisfies the
general requirements listed in section 3. In subsection 4.5, the
document looks at REQ-7 in order to hide the fact that splicing take
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 main RTP stream into 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 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,
and then sends the output RTP media stream to receiver. The mixer
may insert the SSRC of substitutive RTP stream into 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 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, and then sends the
output media stream to the receivers. Moreover, the mixer may insert
the SSRC of main RTP stream into CSRC list in the output media stream
as before.
Note that if the content is too large to fit into RTP packets sent to
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 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, 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 sender and receiver into two separate
RTCP loops, RTP sender has no idea about the situation on the
receiver. But splicing is a processing that 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 downstream receiver), this enables the main RTP sender or
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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 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 main RTP stream) during non-splicing unless the mixer
fragment 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 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 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 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 the
content reaches the mixer when the content is not sent to the RTP
receiver. 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 above RTCP operating mechanism, the RTP sender whose content
is being passed to receiver will see the reception quality of its
stream as received by the mixer, and the reception quality of spliced
stream as received by the receiver. The RTP sender whose content is
not being passed to receiver will only see the reception quality of
its stream as received by the mixer.
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The mixer must forward RTCP 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/AVPF [RFC4585] transport layer
feedback packet, it must handle it carefully as the feedback packet
may contain the information of the content that come 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 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 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
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 the media substitution
is erased at the higher layers are 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
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
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straightforward approach is that the mixer performs an ungraceful
action, terminating the splicing and switching back to main RTP
stream even if this may cause media stuttering on 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 exists,
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
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 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 4.2.
2. If mixer receives the RTCP NACK packets defined in [RFC4585] from
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 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 substitutive RTP sender will directly fetch
the lost substitutive content and retransmit it to RTP receiver.
The mixer may buffer the sent RTP packets and do the
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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 (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 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 bring undesirable delay. Fortunately in this memo, the mixer
acting as 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 DCCP
congestion control algorithms 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 above analysis, to reduce the risk of congestion and remain the
bandwidth consumption stable over time, the substitutive RTP stream
is recommended to be encoded at an appropriate bitrate to match that
of main RTP stream. If the substitutive RTP stream comes from the
substitutive RTP sender, this sender had better has some knowledge
about the media encoding bitrate of main content in advance. How it
knows that is out of scope in this draft.
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
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absence of CSRC list in the output RTP stream, the RTP receiver only
initiates SDES, BYE and APP packets to the mixer without any
knowledge of the main RTP sender and the substitutive RTP sender.
CSRC list identifies the contributing sources, these SSRC identifiers
of contributing sources are kept globally unique for each RTP
session. The uniqueness of SSRC identifier is used to resolve
collisions and detecting RTP-level forwarding loops as defined in
section 8.2 of [RFC3550]. The absence of CSRC list in this case will
create a danger that loops involving those contributing sources could
not be detected. 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 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, 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 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.
If undetectable splicing requirement is required, CSRC list is not
included in 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 or DTLS will use a
security association between the splicer and the receiver. When
using SRTP the splicer could be provisioned with the same security
association as the main RTP sender. Using a limitation in the SRTP
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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 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
senders and the splicer, are not end-to-end. 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 possibility 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 specify or mark where the splicing points exist in the media
streams. When using RTP payload security mechanisms SRTP or other
security mechanism 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
Oran, Cullen Jennings, Ali C Begen, Charles Eckel and Ning Zong.
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9. 10. Appendix- Why Mixer Is Chosen
Translator and mixer both can realize splicing by changing a set of
RTP parameters.
Translator has no SSRC, hence it is transparent to RTP sender and
receiver. Therefore, RTP sender sees the full path to the receiver
when translator is passing its content. When translator insert the
substitutive content RTP sender could get a report on the path up to
translator itself. Additionally, if splicing does not occur yet,
translator does not need to rewrite RTP header, the overhead on
translator can be avoided.
If mixer is used to do splicing, it can also allow RTP sender to
learn the situation of its content on receiver or on mixer just like
translator does, which is specified in section 4.2. Compared to
translator, mixer's outstanding benefit is that it is pretty straight
forward to do with RTCP messages, for example, bit-rate adaptation to
handle varying network conditions. But translator needs more
considerations and its implementation is more complex.
From above analysis, both translator and mixer have their own
advantages: less overhead or less complexity on handling RTCP.
Through long and sophisticated discussion, the avtext WG members
prefer less complexity rather than less overhead and incline to mixer
to do splicing.
If one chooses mixer as splicer, the overhead on mixer must be taken
into account even if the splicing does not occur 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.
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Internet-Draft RTP splicing November 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: xiajinwei@huawei.com
Xia Expires May 18, 2013 [Page 16]
