AVTEXT Working Group J. Xia
Internet-Draft Huawei
Intended status: Informational October 8, 2011
Expires: April 10, 2012
Content Splicing for RTP Sessions
draft-ietf-avtext-splicing-for-rtp-00
Abstract
This memo outlines RTP splicing. Splicing is a process that replaces
the content of the main multimedia stream with other multimedia
content, and delivers the substitutive multimedia content to receiver
for a period of time. This memo provides some RTP splicing use
cases, then we enumerate a set of requirements and analyze whether an
existing RTP level middlebox can meet these requirements, at last we
provide concrete guidelines for how the chosen middlebox works to
handle RTP splicing.
Status of this Memo
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This Internet-Draft will expire on February 12, 2012.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. RTP Splicing Discussion and Requirements . . . . . . . . . . . 5
4. Recommended Solution for RTP Splicing . . . . . . . . . . . . 7
4.1. RTP Processing in RTP Mixer . . . . . . . . . . . . . . . 7
4.2. RTCP Processing in RTP Mixer . . . . . . . . . . . . . . . 9
4.3. Media Clipping Considerations . . . . . . . . . . . . . . 10
4.4. Congestion Control Considerations . . . . . . . . . . . . 11
4.5. Processing Splicing in User Invisibility Case . . . . . . 13
5. Implementation Considerations . . . . . . . . . . . . . . . . 13
6. Security Considerations . . . . . . . . . . . . . . . . . . . 13
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 13
8. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 14
9. Change Log . . . . . . . . . . . . . . . . . . . . . . . . . . 14
9.1. draft-xia-avtext-splicing-for-rtp-01 . . . . . . . . . . . 14
9.2. draft-xia-avtext-splicing-for-rtp-00 . . . . . . . . . . . 14
10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 15
10.1. Normative References . . . . . . . . . . . . . . . . . . . 15
10.2. Informative References . . . . . . . . . . . . . . . . . . 15
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 16
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1. Introduction
This document outlines how splicing can be used for RTP sessions.
Splicing is a process that replaces the content of the main RTP
stream with other multimedia content, and delivers the substitutive
content to receiver for a period of time. The substitutive content
can be provided for example via another RTP stream or local media
file storage.
One representative use case for splicing is advertisements insertion,
which allows operators to replace the national advertising content
with its own regional advertising content prior to delivering the
regional advertising content to receiver.
Besides the advertisement insertion use case, there are other use
cases to which RTP splicing technology can apply. For example,
splicing a recorded video into a video conferencing session, and
implementing a playlist server that stitches pieces of video together
and so forth.
So far [SCTE30] and [SCTE35] have standardized MPEG2-TS splicing
running over cable. The introduction of multimedia splicing into
internet requires changes to transport layer, but to date there is no
guideline for how to handle content splicing for RTP sessions
[RFC3550].
In this document, we first describe a set of requirements of RTP
splicing. Then we provide a method about how an intermediary node
can be used to process RTP splicing to meet these requirements from
the aspects of feasibility, implementation complexity and backward
compatibility.
2. Terminology
The keywords "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
Current RTP Stream
The RTP stream that the RTP receiver is currently receiving. The
content of current RTP stream can be either main content or
substitutive content.
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Main Content
The multimedia content that are conveyed in 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 main
RTP stream can be replaced by substitutive content for a period of
time.
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 that may provide substitutive content. 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 Splicer before the
substitutive content is delivered to receiver.
Splicing In Point
A virtual point in the RTP stream, suitable for substitutive
content entry, that exists in the boundary of two independently
decodable frames.
Splicing Out Point
A virtual point in the RTP stream, suitable for substitutive
content exist, that exists in the boundary of two independently
decodable frames.
Splicer
An intermediary node that inserts substitutive content into main
RTP stream. Splicer sends substitutive content to RTP receiver
instead of main content during splicing. It is also responsible
for processing RTCP traffic between media source and RTP receiver.
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3. RTP Splicing Discussion and Requirements
In this document, we assume an intermediary network element, which is
referred to as Splicer, to play the key role to handle RTP splicing.
A simplified RTP splicing diagram is depicted in Figure 1, in which
only one main content flow and one substitutive content flow are
given.
+---------------+
| | Main Content +-----------+
|Main RTP Sender|------------->| | Current Content
| | | Splicer |---------->
+---------------+ ---------->| |
| +-----------+
|
| Substitutive Content
|
|
+-----------------------+
|Substitutive RTP Sender|
| or |
| Local File Storage |
+-----------------------+
Figure 1: RTP Splicing Architecture
When RTP splicing begins, Splicer stops delivering the main content,
instead delivering the substitutive content to RTP receiver for a
period of time, and then resumes the main content when splicing ends.
The methods how Splicer learns when to start and end the splicing is
out of scope for this document. The RTP splicing may happen more
than once in case that substitutive content will be dispersedly
inserted in multiple time slots during the lifetime of the main RTP
stream.
When realizing splicing technology on RTP layer, there are a set of
requirements that must be satisfied to at least some degree on
Splicer:
REQ-1:
Splicer MUST operate in either unicast or multicast session
environment.
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REQ-2:
Splicer SHOULD NOT cause perceptible media clipping at the
splicing point and adverse impact on the quality of user
experience.
REQ-3:
Splicer MUST be backward compatible with RTP/RTCP protocols, and
its associated profiles and extensions to those protocols. For
example, Splicer MUST be robust to packet loss, network congestion
etc.
REQ-4:
Splicer MUST be trusted by media source and receiver, and has the
valid security context with media source and RTP receiver
respectively.
REQ-5:
Splicer SHOULD allow the media source to learn the performance of
the downstream receiver when its content is being passed to RTP
receiver.
In a number of deployment scenarios, especially advertisement
insertion, there may be one specific requirement. Given that it is
unacceptable for advertisers that their advertising content is not
delivered to user, this may require RTP splicing to be operated
within the following constraint:
If Splicer intends to prevent RTP receiver from identifying and
filtering the substitutive content, it SHOULD eliminate the
visibility of splicing process on RTP level from RTP receiver
point of view.
However, substitutive content and main content are encoded by
different encoders and have different parameter sets. In such
case, a full media transcoding must be done on Splicer to ensure
the completely invisible impact on RTP receiver, but this may be
prohibitively expensive and complex. As a trade-off, it is
RECOMMENDED to minimize the splicing visibility on RTP receiver,
i.e., maintaining RTP header parameters consistent but leaving the
RTP payload untranscoded. If one wants to realize complete
invisibility, the cost of transcoding must be taken into account.
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Henceforth, we refer to the minimum and complete invisibility
requirement as User Invisibility Requirement.
To improve the versatility of existing implementations and better
interoperability, it is RECOMMENDED to use existing tools in RTP/RTCP
protocol family to realize RTP splicing without any protocol
extension unless the existing tools are incompetent for splicing.
4. Recommended Solution for RTP Splicing
Given that Splicer is an intermediary node exists between the main
media source and the RTP receiver and splicing is not an very
complicated processing, there are some chance that any existing RTP-
level middlebox may has the incidental capability to meet the
requirements described in previous section.
Since Splicer switches between substitutive content and main content,
and only forward one of them at one point of time. A RFC3550 mixer
seems to have such capability to select a media stream, always under
its own SSRC, which can be used for reception reports to media
sources and receivers. Moreover, mixer includes the CSRC list in
outgoing packets to indicate the source(s) of content, this
facilitates the system debugging. From this point of view, an RTP
mixer may have some chance to be Splicer. In next four subsections
(from subsection 4.1 to subsection 4.4), we start analyzing how an
RTP mixer handles RTP splicing and how it satisfies the general
requirements listed in section 3.
In subsection 4.5, we specially consider the special requirement 6
(i.e., User Invisibility Requirement) since it needs to mask any
splicing clue on user (e.g, CSRC list must not be included in
outgoing packets to prevent user from identifying the difference
between main RTP stream and substitutive RTP stream) when mixer is
used.
4.1. RTP Processing in RTP Mixer
Once mixer has learnt when to do splicing, it must get ready for the
coming splicing in advance, e.g., fetches the substitutive content
either from local media file storage or via substitutive RTP stream
earlier than splicing in point. If the substitutive content comes
from local media file storage, mixer can construct the substitutive
RTP stream using its own SSRC and leave the CSRC list blank in the
output stream.
When the main RTP stream begins, mixer terminates the main RTP
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stream. Using the main RTP packets, mixer generates the current
media stream with its own SSRC, sequence number space and timing
model. Moreover, mixer inserts the SSRC of main RTP stream into CSRC
list in the current media stream.
When splicing begins, mixer switches to the substitutive RTP stream
at splicing in point, extracts the payload data (i.e., substitutive
content), encodes substitutive content and outputs it instead of main
content in the current media stream. Moreover, mixer inserts the
SSRC of substitutive RTP stream into CSRC list in the current media
stream.
When splicing ends, mixer switches to the main RTP stream at splicing
out point, extracts the payload data (i.e., main content), encodes
main content and outputs it instead of substitutive content in the
current media stream. Moreover, mixer inserts the SSRC of main RTP
stream into CSRC list in the current media stream.
The whole RTP splicing procedure is perhaps best explained by a
pseudo code example:
if (main RTP stream begins) {
the main RTP stream is terminated on mixer and main content is
encoded by mixer with its own SSRC identifier;
the sequence numbers of the current RTP packets which contain main
content are allocated by mixer, until the splicing begins;
the timestamp of the current RTP packet increments linearly;
the CSRC list of the current RTP packet indicates SSRC of main RTP
stream;
}
else if (splicing begins) {
the substitutive RTP stream is terminated on mixer and
substitutive content is encoded by mixer with its own SSRC
identifier;
the sequence numbers of the current RTP packets which contain
substitutive content are allocated by mixer and maintain
consistent with the sequence numbers of previous current RTP
packets, until the splicing end;
the timestamp of the current RTP packet increments linearly;
the CSRC list of the current RTP packet indicates SSRC of
substitutive RTP stream;
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}
else if (splicing ends) {
the main RTP stream is terminated on mixer and main content is
encoded by mixer with its own SSRC identifier;
the sequence numbers of the current RTP packets which contain main
content are allocated by mixer and maintain consistent with the
sequence numbers of previous current RTP packets, until the next
splicing begins;
the timestamp of the current RTP packets increments linearly;
the CSRC list the current RTP indicates SSRC of main RTP stream;
}
Splicing may occur more than one time during the lifetime of main RTP
stream, this means mixer needs to output main content and
substitutive content in turn with its own SSRC identifier. From user
point of view, the only source of the current stream is mixer
wherever the content comes from.
Note that, the substitutive content should be outputted in the range
of splicing duration. Any gap or overlap between main RTP stream and
substitutive RTP stream may induce media clipping at splicing point.
More details about preventing media clipping are introduced in
section 4.3.
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 situation on it and can communicate this
information to media source via RTCP reception reports.
According to the description in section 7.3 of [RFC3550], mixer
divides RTCP flow between media source and receiver into two separate
RTCP loops, media source probably has no idea about the situation on
receiver. Hence, mixer may use some mechanisms, allowing media
source to at least some degree to have some knowledge of the
situation on receiver when its content is being passed to receiver.
Because splicing is a processing that mixer selects one media stream
from multiple streams rather than mixing them, the number of the
original, pre-spliced RTP packets will be equal to the number of
spliced RTP packets, i.e., mixer changes the sequence numbers of RTP
packets but not changes the sum of them. When mixer receives RTCP
reception reports sent from downstream receiver, it must change the
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SSRC in report block to the SSRC identifier of original media source
and rewrite the extended highest sequence number to the corresponding
original extended highest sequence number before forwarding the RTCP
reception reports to original media source. In such case, the media
source will see the reception quality of its stream received by
mixer, and the reception quality of spliced stream received by RTP
receiver.
For the media source whose content is terminated on mixer and is not
being passed to receiver, mixer must act as a receiver and therefore
send reception reports to the media source.
4.3. Media Clipping Considerations
This section provides informative guideline about how media clipping
may shape and how mixer deal with the media clipping.
Media clippings potentially occur at the splicing point if the time
slot for substitutive RTP stream mismatches (shorter or longer than)
the duration of the reserved main RTP stream for replacing.
At the splicing in point, mixer usually start the substitutive
content with independently decodable frames and fill the substitutive
content up receiver's buffer with several seconds earlier than the
presentation time of substitutive content. By this way, smooth
playback can be achieved without pauses or stuttering.
Compared to buffering method used at splicing in point, things become
somewhat complex at splicing out point. The case that insertion
duration is shorter than the reserved gap time may cause a little
playback latency of main RTP stream on RTP receiver, but not
adversely impact the quality of user experience. However, in case
that insertion duration is longer than the reserved gap duration,
there exists an overlap of the substitutive RTP stream and the main
RTP stream at splicing out point, which may cause synchronization
problem and result in a perceptible media clipping.
To guard against a media clipping at splicing out point, main RTP
source may reserve a bit extra playback delay (e.g., 500
milliseconds) to send out the first main RTP packet after splicing
ends. Note that the delay should not be too long to smoothly
playback the coming main RTP stream. But if the splicing is still
unfinished when the first main RTP packet has reached, mixer must
terminate the splicing and switch back to main RTP stream even if
this may cause media stuttering on receiver.
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4.4. Congestion Control Considerations
Provided that the substitutive content has somewhat different
characteristics to the main content it replaces (e.g., the more
dynamic content, the higher bandwidth occupation), or substitutive
content may be encoded with different codec and has different
encoding bitrate, some challenge raise to network capacity and
receiver buffer size. A more dynamic content or a higher encoding
bitrate stream might overload the network and possibly exceed the
receiver's media consumption rate, which might flood receiver's
buffer and eventually result in a buffer overflow. Either network
overload or buffer overflow would induce network congestion and
congestion-caused packet loss.
To be robust to network congestion and packet loss, mixer must
continuously monitor the network situation by means of a variety of
manners:
1. RTCP receiver reports indicate packet loss [RFC3550].
2. RTCP NACKs for lost packet recovery [RFC4585].
3. RTCP ECN Feedback information [I-D.ietf-avtcore-ecn-for-rtp].
Upon detection of above three types of RTCP reports during splicing,
mixer will treat them with three different manners as following:
1. If mixer receives the RTCP receiver reports with packet loss
indication, it will process them as the description given in
section 7.3 of [RFC3550].
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, 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, mixer
acting as substitutive media source will directly fetch the lost
substitutive content and retransmit it to RTP receiver.
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, 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.
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3. In [I-D.ietf-avtcore-ecn-for-rtp], two RTCP extensions are
defined for ECN feedback: RTP/AVPF transport layer ECN feedback
packet for urgent ECN information, and RTCP XR ECN summary report
block for regular reporting of the ECN marking information.
If an ECN-aware mixer receives any RTCP ECN feedback (i.e., RTCP
ECN feedback packets or RTCP XR summary reports) from RTP
receiver, it must operates as description given in section 8.4 of
[I-D.ietf-avtcore-ecn-for-rtp], terminating the RTCP ECN feedback
packets from downstream receivers, and driving congestion control
loop and bitrate adaptation between itself and downstream
receiver as if it were the media source. In addition, an ECN-
aware RTP mixer must generate RTCP ECN feedback relating to the
input RTP streams it terminates, and driving congestion control
loop and bitrate adaptation between itself and upstream sender as
if it were the RTP sender.
Once mixer learns that congestion is being experienced on its
downstream link by means of above three detection mechanisms, it
should adapt the bitrate of output stream in response to network
congestion. The bitrate adaptation may be determined by a TCP-
friendly bitrate adaptation algorithm specified in [RFC5348], or by a
DCCP congestion control algorithms defined in [RFC5762].
In practice, during splicing, the real reason to cause congestion
usually is the different characteristic of substitutive RTP stream
(more dynamic content or higher encoding bitrate) with main RTP
stream, and that stream transcoding or thinning on mixer is very
inefficient and difficult operation. Therefore, a means that enables
substitutive media source to limit the media bitrate it is currently
generating even in the absence of congestion on the path between
itself and mixer is desirable. The TMMBR message defined in
[RFC5104] provides an effective method. When mixer detects
congestion on its downstream link during splicing, it uses TMMBR to
request substitutive media source to reduce the media bitrate to a
value that is in compliance with congestion control principles for
the slowest link. Upon reception of TMMBR, substitutive media source
applies its congestion control algorithm and responds Temporary
Maximum Media Stream Bit Rate Notification (TMMBN) to mixer.
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
substitutive media source, the source 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.
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4.5. Processing Splicing in User Invisibility Case
Compared to above user visibility case, the primary difference in
this case is mixer MUST NOT include CSRC list in outgoing packets
(i.e., CSRC count field is set to zero and CSRC list fields are
absent).
Therefore, due to the absence of CRSC list in current RTP stream, RTP
receiver only initiates SDES, BYE and APP packets to mixer without
any knowledge of main media source and substitutive media source.
This creates a danger that loops involving those sources could not be
detected.
5. Implementation Considerations
When 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 mixer which uses its own sequence number
space and timing model. So mixer will rewrite RTP sequence number
and timestamp whatever splicing is active or not, and generate RTCP
flows for both sides. In case mixer serves multiple main RTP streams
simultaneously, this may lead to more overhead on mixer.
In addition, there is a potential issue with loop detection, which
would be problematic if User Invisibility Requirement is required.
6. Security Considerations
If any payload internal security mechanisms (e.g., SSH, SSL etc) are
used, only media source and RTP receiver can learn the security
keying material generated by such internal security mechanism, any
middlebox (e.g., mixer) between media source and RTP receiver can't
get such keying material. Only when regular transport security
mechanisms (e.g., SRTP, IPSec, etc) are used, mixer will process the
packets passing through it.
The security considerations of the RTP specification [RFC3550], the
Extended RTP profile for RTCP-Based Feedback [RFC4585], and the
Secure Real-time Transport Protocol [RFC3711] apply. Mixer must be
trusted by main media source and insertion media source, and must be
included in the security context.
7. IANA Considerations
No IANA actions are required.
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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, and Ning Zong.
9. Change Log
9.1. draft-xia-avtext-splicing-for-rtp-01
The following are the major changes compared to previous version 00:
o Use mixer to handle both user visible and invisible splicing.
o Add one subsection to describe media clipping considerations.
o Add one subsection to describe congestion control considerations.
9.2. draft-xia-avtext-splicing-for-rtp-00
The following are the major changes compared to previous AVT I-D
version 00:
o Change primary RTP stream to main RTP stream, add current RTP
stream as the streaming received by RTP receiver.
o Eliminate the ambiguity of inserted content with substitutive
content which replaces the main content rather than pause it.
o Clarify the signaling requirements.
o Delete the description on Mixer and MCU in section 4, mainly focus
on the direction whether a Translator can act as a Splicer.
o Add section 5 to describe the exact guidance on how an RTP
Translator is used to handle splicing.
o Modify the security considerations section and add acknowledges
section.
10. References
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10.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2250] Hoffman, D., Fernando, G., Goyal, V., and M. Civanlar,
"RTP Payload Format for MPEG1/MPEG2 Video", RFC 2250,
January 1998.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[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.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, February 2008.
[RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117,
January 2008.
[I-D.ietf-avtcore-ecn-for-rtp]
Westerlund, M., "Explicit Congestion Notification (ECN)
for RTP over UDP", draft-ietf-avtcore-ecn-for-rtp-02 (work
in progress), October 2010.
10.2. Informative References
[RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification",
RFC 5348, September 2008.
[RFC5760] Ott, J., Chesterfield, J., and E. Schooler, "RTP Control
Protocol (RTCP) Extensions for Single-Source Multicast
Sessions with Unicast Feedback", RFC 5760, February 2010.
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[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", 2001.
[SCTE35] Society of Cable Telecommunications Engineers (SCTE),
"Digital Program Insertion Cueing Message for Cable",
2004.
[H.323] ITU-T Recommendation H.323, "Packet-based multimedia
communications systems", June 2006.
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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