Internet DRAFT - draft-ietf-avtext-splicing-notification
draft-ietf-avtext-splicing-notification
AVTEXT Working Group J. Xia
INTERNET-DRAFT R. Even
Intended Status: Standards Track R. Huang
Expires: February 4, 2017 Huawei
L. Deng
China Mobile
August 3, 2016
RTP/RTCP extension for RTP Splicing Notification
draft-ietf-avtext-splicing-notification-09
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. The splicer is designed to handle RTP splicing and needs to
know when to start and end the splicing.
This memo defines two RTP/RTCP extensions to indicate the splicing
related information to the splicer: an RTP header extension that
conveys the information in-band and an RTCP packet that conveys the
information out-of-band.
Status of this Memo
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Copyright and License Notice
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Table of Contents
1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1 Terminology . . . . . . . . . . . . . . . . . . . . . . . . 3
2 Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
2.1 Overview of RTP Splicing . . . . . . . . . . . . . . . . . . 4
2.2 Overview of Splicing Interval . . . . . . . . . . . . . . . 5
3 Conveying Splicing Interval in RTP/RTCP extensions . . . . . . 5
3.1 RTP Header Extension . . . . . . . . . . . . . . . . . . . . 5
3.2 RTCP Splicing Notification Message . . . . . . . . . . . . . 6
4 Reducing Splicing Latency . . . . . . . . . . . . . . . . . . . 7
5 Failure Cases . . . . . . . . . . . . . . . . . . . . . . . . . 8
6 SDP Signaling . . . . . . . . . . . . . . . . . . . . . . . . . 8
6.1 Declarative SDP . . . . . . . . . . . . . . . . . . . . . . 9
6.2 Offer/Answer without BUNDLE . . . . . . . . . . . . . . . . 9
6.3 Offer/Answer with BUNDLE: All Media are spliced . . . . . . 10
6.4 Offer/Answer with BUNDLE: a Subset of Media are Spliced . . 12
7 Security Considerations . . . . . . . . . . . . . . . . . . . . 13
8 IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 14
8.1 RTCP Control Packet Types . . . . . . . . . . . . . . . . . 14
8.2 RTP Compact Header Extensions . . . . . . . . . . . . . . . 14
8.3 SDP Grouping Semantic Extension . . . . . . . . . . . . . . 14
9 Acknowledges . . . . . . . . . . . . . . . . . . . . . . . . . . 15
10 References . . . . . . . . . . . . . . . . . . . . . . . . . . 15
10.1 Normative References . . . . . . . . . . . . . . . . . . . 15
10.2 Informative References . . . . . . . . . . . . . . . . . . 15
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 16
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1 Introduction
Splicing is a process that replaces some multimedia content with
other multimedia content and delivers the substitutive multimedia
content to the receivers for a period of time. In some predictable
splicing cases, e.g., advertisement insertion, the splicing duration
needs to be inside of the specific, pre-designated time slot. Certain
timing information about when to start and end the splicing must be
first acquired by the splicer in order to start the splicing. This
document refers to this information as the Splicing Interval.
[SCTE35] provides a method that encapsulates the Splicing Interval
inside the MPEG2-TS layer in cable TV systems. When transported in
RTP, an middle box designed as the splicer to decode the RTP packets
and search for the Splicing Interval inside the payloads is required.
The need for such processing increases the workload of the middle box
and limits the number of RTP sessions the middle box can support.
The document defines an RTP header extension [RFC5285bis] used by the
main RTP sender to provide the Splicing Interval by including it in
the RTP packets.
However, the Splicing Interval conveyed in the RTP header extension
might not reach the splicer successfully. Any splicing un-aware
middlebox on the path between the RTP sender might strip this RTP
header extension.
To increase robustness against such case, the document also defines a
new RTCP packet type to carry the same Splicing Interval to the
splicer. Since RTCP is also unreliable and may not be so immediate as
the in-band way, it's only considered as a complement to the RTP
header extension.
1.1 Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
In addition, we define following terminologies:
Main RTP sender:
The sender of RTP packets carrying the main RTP stream.
Splicer:
An intermediary node that inserts substitutive content into a main
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RTP stream. The splicer sends substitutive content to the RTP
receiver instead of the main content during splicing. It is also
responsible for processing RTCP traffic between the RTP sender and
the RTP receiver.
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.
Splicing Interval:
The NTP-format timestamps, representing the main RTP sender
wallclock time, for the Splicing-In point and Splicing-Out point
per [RFC6828] allowing the splicer to know when to start and end
the RTP splicing.
Substitutive RTP Sender:
The sender of RTP packets carrying the RTP stream that will
replace the content in the main RTP stream.
2 Overview
2.1 Overview of RTP Splicing
RTP Splicing is intended to replace some multimedia content with
certain substitutive multimedia content, and then forward it to the
receivers for a period of time. This process is authorized by the
main RTP sender that offers a specific time window for inserting the
substitutive multimedia content in the main content. A typical usage
is that IPTV service provider uses its own regional advertising
content to replace national advertising content, the time window of
which is explicitly indicated by the IPTV service provider.
The splicer is a middlebox handling RTP splicing. It receives main
content and substitutive content simultaneously but only chooses to
send one of them to the receiver at any point of time. When RTP
splicing begins, the splicer sends the substitutive content to the
receivers instead of the main content. When RTP splicing ends, the
splicer switches back to sending the main content to the receivers.
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This implies that the receiver is explicitly configured to receive
the traffic via the splicer, and will return any RTCP feedback to it
in the presence of the splicer.
The middlebox working as the splicer can be implemented as either an
RTP mixer or as an RTP translator. If implemented as an RTP mixer,
the splicer will use its own SSRC, sequence number space, and timing
model when generating the output stream to receivers, using the CSRC
list to indicate whether the original or substitutive content is
being delivered. The splicer, on behalf of the content provider, can
omit the CSRC list from the RTP packets it generates. This simplifies
the design of the receivers, since they don't need to parse the CSRC
list, but makes it harder to determine when the splicing is taking
place (it requires inspection of the RTP payload data, rather than
just the RTP headers). A splicer working as an RTP mixer splits the
flow between the sender and receiver into two, and requires separate
control loops, for RTCP and congestion control. [RFC6828] offers an
example of an RTP mixer approach.
A splicer implemented as an RTP translator [RFC3550] will forward the
RTP packets from the original and substitutive senders with their
SSRCs intact, but will need to rewrite RTCP sender report packets to
account for the splicing. In this case, the congestion control loops
run between original sender and receiver, and between the
substitutive sender and receiver. The splicer needs to ensure that
the RTCP feedback message from the receiver are passed to the right
sender to let the congestion control work.
2.2 Overview of Splicing Interval
To handle splicing on the RTP layer at the reserved time slots set by
the main RTP sender, the splicer must first know the Splicing
Interval from the main RTP sender before it can start splicing.
When a new splicing is forthcoming, the main RTP sender needs to send
the Splicing Interval to the splicer. The Splicing Interval SHOULD be
sent by RTP header extension or RTCP extension message more than once
to mitigate the possible packet loss. To enable the splicer to get
the substitutive content before the splicing starts, the main RTP
sender MUST send the Splicing Interval far ahead. For example, the
main RTP sender can estimate when to send the Splicing Interval based
on the round-trip time (RTT) following the mechanisms in section
6.4.1 of [RFC3550] when the splicer sends RTCP RR to the main sender.
The substitutive sender also needs to learn the Splicing Interval
from the main RTP sender in advance, and thus estimates when to
transfer the substitutive content to the splicer. The Splicing
Interval could be transmitted from the main RTP sender to the
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substitutive content using some out-of-band mechanisms, for example,
a proprietary mechanism to exchange the Splicing Interval, or the
substitutive sender is implemented together with the main RTP sender
inside a single device. To ensure the Splicing Interval is valid for
both the main RTP sender and the substitutive RTP sender, the two
senders MUST share a common reference clock so that the splicer can
achieve accurate splicing. The requirements for the common reference
clock (e.g. resolution, skew) depend on the codec used by the media
content.
In this document, the main RTP sender uses a pair of NTP-format
timestamps, to indicate when to start and end the splicing to the
splicer: the timestamp of the first substitutive RTP packet at the
splicing in point, and the timestamp of the first main RTP packet at
the splicing out point.
When the substitutive RTP sender gets the Splicing Interval, it must
prepare the substitutive stream. The main and the substitutive
content providers MUST ensure that the RTP timestamp of the first
substitutive RTP packet that would be presented to the receivers
corresponds to the same time instant as the former NTP-format
timestamp in the Splicing Interval. To enable the splicer to know the
first substitutive RTP packet it needs to send, the substitutive RTP
sender MUST send the substitutive RTP packet ahead of the Splicing In
point, allowing the splicer to find out the timestamp of this first
RTP packet in the substitutive RTP stream, e.g., using a prior RTCP
SR (Sender Report) message.
When the splicing will end, the main content provider and the
substitutive content provider MUST ensure the RTP timestamp of the
first main RTP packet that would be presented on the receivers
corresponds to the same time instant as the latter NTP-format
timestamp in the Splicing Interval.
3 Conveying Splicing Interval in RTP/RTCP extensions
This memo defines two backwards compatible RTP extensions to convey
the Splicing Interval to the splicer: an RTP header extension and an
RTCP splicing notification message.
3.1 RTP Header Extension
The RTP header extension mechanism defined in [RFC5285bis] can be
adapted to carry the Splicing Interval consisting of a pair of NTP-
format timestamps.
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This RTP header extension carries the 7 octets splicing-out NTP-
format timestamp (lower 24-bit part of the Seconds of a NTP-format
timestamp and the 32 bits of the Fraction of a NTP-format timestamp
as defined in [RFC5905]), followed by the 8 octets splicing-in NTP-
format timestamp (64-bit NTP-format timestamp as defined in
[RFC5905]). The top 8 bits of the splicing-out NTP timestamp are
inferred from the top 8 bits of the splicing-in NTP timestamp, under
the assumption that the splicing-out time is after the splicing-in
time, and the splicing interval is less than 2^25 seconds. Therefore,
if the value of 7 octets splicing-out NTP-format timestamp is smaller
than the value of 7 lower octets splicing-in NTP-format, it implies a
wrap of the 56-bit splicing-out NTP-format timestamp which means the
top 8-bit value of the 64-bit splicing-out is equal to the top 8-bit
value of splicing-in NTP Timestamp plus 0x01. Otherwise, the top 8
bits of splicing-out NTP timestamp is equal to the top 8 bits of
splicing-in NTP Timestamp.
This RTP header extension can be encoded using either the one-byte or
two-byte header defined in [RFC5285bis]. Figure 1 and 2 show the
splicing interval header extension with each of the two header
formats.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+E
| ID | L=14 | OUT NTP timestamp format - Seconds (bit 8-31) |x
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+t
| OUT NTP timestamp format - Fraction (bit 0-31) |e
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+n
| IN NTP timestamp format - Seconds (bit 0-31) |s
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+i
| IN NTP timestamp format - Fraction (bit 0-31) |o
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+n
Figure 1: Splicing Interval
Using the One-Byte Header Format
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+E
| ID | L=15 | OUT NTP timestamp - Seconds |x
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+t
|Out Secds(cont)| OUT NTP timestamp format - Fraction |e
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+n
|Out Fract(cont)| IN NTP timestamp format - Seconds |s
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+i
| In Secds(cont)| IN NTP timestamp format - Fraction |o
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+n
| In Fract(cont)| 0 (pad) | ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 2: Splicing Interval
Using the Two-Byte Header Format
Since the inclusion of an RTP header extension will reduce the
efficiency of RTP header compression, it is RECOMMENDED that the main
sender inserts the RTP header extensions into only a number of RTP
packets, instead of all the RTP packets, prior to the splicing in.
After the splicer obtains the RTP header extension and derives the
Splicing Interval, it generates its own stream and is not allowed to
include the RTP header extension in outgoing packets to reduce header
overhead.
3.2 RTCP Splicing Notification Message
In addition to the RTP header extension, the main RTP sender includes
the Splicing Interval in an RTCP splicing notification message.
Whether or not the timestamps are included in the RTP header
extension, the main RTP sender MUST send the RTCP splicing
notification message. This provide robustness in the case where a
middlebox strips RTP header extensions. The main RTP sender MUST make
sure the splicing information contained in the RTCP splicing
notification message consistent with the information included in the
RTP header extensions.
The RTCP splicing notification message is a new RTCP packet type. It
has a fixed header followed by a pair of NTP-format timestamps:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|reserved | PT=TBA | length |
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+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| IN NTP Timestamp (most significant word) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| IN NTP Timestamp (least significant word) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| OUT NTP Timestamp (most significant word) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| OUT NTP Timestamp (least significant word) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 2: RTCP Splicing Notification Message
The RSI packet includes the following fields:
Length: 16 bits
As defined in [RFC3550], the length of the RTCP packet in 32-bit
words minus one, including the header and any padding.
SSRC: 32 bits
The SSRC of the Main RTP Sender.
Timestamp: 64 bits
Indicates the wallclock time when this splicing starts and ends.
The full-resolution NTP-format timestamp is used, which is a 64-
bit, unsigned, fixed-point number with the integer part in the
first 32 bits and the fractional part in the last 32 bits. This
format is same as the NTP timestamp field in the RTCP Sender
Report (Section 6.4.1 of [RFC3550]).
The RTCP splicing notification message can be included in the RTCP
compound packet together with RTCP SR generated at the main RTP
sender, and hence follows the compound RTCP rules defined in Section
6.1 in [RFC3550].
If the use of non-compound RTCP [RFC5506] was previously negotiated
between the sender and the splicer, the RTCP splicing notification
message may be sent as non-compound RTCP packets. In some cases that
the mapping from RTP timestamp to NTP timestamp changes, e.g., clock
drift happening before the splicing event, it may be required to send
RTCP SR or even updated Splicing Interval information timely to
update the timestamp mapping for accurate splicing.
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Since the RTCP splicing notification message is intentionally sent by
the main RTP sender to the splicer, the splicer is not allowed to
forward this message to the receivers so as to avoid their useless
processing and additional RTCP bandwidth consumption in the
downstream.
4 Reducing Splicing Latency
When splicing starts or ends, the splicer outputs the multimedia
content from another sender to the receivers. Given that the
receivers must first acquire certain information ([RFC6285] refers to
this information as Reference Information) to start processing the
multimedia data, either the main RTP sender or the substitutive
sender SHOULD provide the Reference Information together with its
multimedia content to reduce the delay caused by acquiring the
Reference Information. The methods by which the Reference Information
is distributed to the receivers is out of scope of this memo.
Another latency element is synchronization caused delay. The
receivers must receive enough synchronization metadata prior to
synchronizing the separate components of the multimedia streams when
splicing starts or ends. Either the main RTP sender or the
substitutive sender SHOULD send the synchronization metadata early
enough so that the receivers can play out the multimedia in a
synchronized fashion. The main RTP sender or the substitutive sender
can estimate when to send the synchronization metadata based on, for
example, the round-trip time (RTT) following the mechanisms in
section 6.4.1 of [RFC3550] when the splicer sends RTCP RR to the main
sender or the substitutive sender. The main RTP sender and the
substitutive sender can also be coordinated by some proprietary out-
of-band mechanisms to decide when and whom to send the metadata. If
both send the information, the splicer SHOULD pick one based on the
current situation, e.g., choosing main RTP sender when synchronizing
the main media content while choosing the information from the
substitutive sender when synchronizing the spliced content. The
mechanisms defined in [RFC6051] are RECOMMENDED to be adopted to
reduce the possible synchronization delay.
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5 Failure Cases
This section examines the implications of losing RTCP splicing
notification message and the other failure case, e.g., the RTP header
extension is stripped on the path.
Given that there may be a splicing un-aware middlebox on the path
between the main RTP sender and the splicer, the main and the
substitutive RTP senders can use one heuristic to verify whether or
not the Splicing Interval reaches the splicer.
The splicer can be implemented to have its own SSRC, and send RTCP
reception reports to the senders of the main and substitutive RTP
streams. This allows the senders to detect problems on the path to
the splicer. Alternatively, it is possible to implement the splicer
such that it has no SSRC, and does not send RTCP reports; this
prevents the senders from being able to monitor the quality to the
path to the splicer.
If the splicer has an SSRC and sends its own RTCP reports, it can
choose not to pass RTCP reports it receives from the receivers to the
senders. This will stop the senders from being able to monitor the
quality of the paths from the splicer to the receivers.
A splicer that has an SSRC can choose to pass RTCP reception reports
from the receivers back to the senders, after modifications to
account for the splicing. This will allow the senders the monitor the
quality of the paths from the splicer to the receivers. A splicer
that does not have its own SSRC has to forward and translation RTCP
reports from the receiver, otherwise the senders will not see any
receivers in the RTP session.
If the splicer is implemented as a mixer, it will have its own SSRC
and will send its own RTCP reports, and will forward translated RTCP
reports from the receivers.
Upon the detection of a failure, the splicer can communicate with the
main sender and the substitutive sender in some out of band signaling
ways to fall back to the payload specific mechanisms it supports,
e.g., MPEG-TS splicing solution defined in [SCTE35], or just abandon
the splicing.
6 Session Description Protocol (SDP) Signaling
This document defines the URI for declaring this header extension in
an extmap attribute to be "urn:ietf:params:rtp-hdrext:splicing-
interval".
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This document extends the standard semantics defined in SDP Grouping
Framework [RFC5888] with a new semantic: SPLICE to represent the
relationship between the main RTP stream and the substitutive RTP
stream. Only 2 m-lines are allowed in the SPLICE group. The main RTP
stream is the one with the extended extmap attribute, and the other
one is substitutive stream. A single m-line MUST NOT be included in
different SPLICE groups at the same time. The main RTP sender
provides the information about both main and substitutive sources.
The extended SDP attribute specified in this document is applicable
for offer/answer content [RFC3264] and do not affect any rules when
negotiating offer and answer. When used with multiple m-lines,
substitutive RTP MUST be applied only to the RTP packets whose SDP m-
line is in the same group with the substitutive stream using SPLICE
and has the extended splicing extmap attribute. This semantic is also
applicable for BUNDLE cases.
The following examples show how SDP signaling could be used for
splicing in different cases.
6.1 Declarative SDP
v=0
o=xia 1122334455 1122334466 IN IP4 splicing.example.com
s=RTP Splicing Example
t=0 0
a=group:SPLICE 1 2
m=video 30000 RTP/AVP 100
i=Main RTP Stream
c=IN IP4 233.252.0.1/127
a=rtpmap:100 MP2T/90000
a=extmap:1 urn:ietf:params:rtp-hdrext:splicing-interval
a=mid:1
m=video 30002 RTP/AVP 100
i=Substitutive RTP Stream
c=IN IP4 233.252.0.2/127
a=sendonly
a=rtpmap:100 MP2T/90000
a=mid:2
Figure 3: Example SDP for a single-channel splicing scenario
The splicer receiving the SDP message above receives one MPEG2-TS
stream (payload 100) from the main RTP sender (with multicast
destination address of 233.252.0.1) on port 30000, and/or receives
another MPEG2-TS stream from the substitutive RTP sender (with
multicast destination address of 233.252.0.2) on port 30002. But at
a particular point in time, the splicer only selects one stream and
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outputs the content from the chosen stream to the downstream
receivers.
6.2 Offer/Answer without BUNDLE
SDP Offer - from main RTP sender
v=0
o=xia 1122334455 1122334466 IN IP4 splicing.example.com
s=RTP Splicing Example
t=0 0
a=group:SPLICE 1 2
m=video 30000 RTP/AVP 31 100
i=Main RTP Stream
c=IN IP4 splicing.example.com
a=rtpmap:31 H261/90000
a=rtpmap:100 MP2T/90000
a=extmap:1 urn:ietf:params:rtp-hdrext:splicing-interval
a=sendonly
a=mid:1
m=video 40000 RTP/AVP 31 100
i=Substitutive RTP Stream
c=IN IP4 substitutive.example.com
a=rtpmap:31 H261/90000
a=rtpmap:100 MP2T/90000
a=sendonly
a=mid:2
SDP Answer - from splicer
v=0
o=xia 1122334455 1122334466 IN IP4 splicer.example.com
s=RTP Splicing Example
t=0 0
a=group:SPLICE 1 2
m=video 30000 RTP/AVP 100
i=Main RTP Stream
c=IN IP4 splicer.example.com
a=rtpmap:100 MP2T/90000
a=extmap:1 urn:ietf:params:rtp-hdrext:splicing-interval
a=recvonly
a=mid:1
m=video 40000 RTP/AVP 100
i=Substitutive RTP Stream
c=IN IP4 splicer.example.com
a=rtpmap:100 MP2T/90000
a=recvonly
a=mid:2
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6.3 Offer/Answer with BUNDLE: All Media are spliced
In this example, the bundled audio and video media have their own
substitutive media for splicing:
1. An Offer, in which the offerer assigns a unique address and a
substitutive media to each bundled "m="line for splicing within the
BUNDLE group.
2. An answer, in which the answerer selects its own BUNDLE address,
and leave the substitutive media untouched.
SDP Offer - from main RTP sender
v=0
o=alice 1122334455 1122334466 IN IP4 splicing.example.com
s=RTP Splicing Example
c=IN IP4 splicing.example.com
t=0 0
a=group:SPLICE foo 1
a=group:SPLICE bar 2
a=group:BUNDLE foo bar
m=audio 10000 RTP/AVP 0 8 97
a=mid:foo
b=AS:200
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=extmap:1 urn:ietf:params:rtp-hdrext:splicing-interval
a=sendonly
m=video 10002 RTP/AVP 31 32
a=mid:bar
b=AS:1000
a=rtpmap:31 H261/90000
a=rtpmap:32 MPV/90000
a=extmap:2 urn:ietf:params:rtp-hdrext:splicing-interval
a=sendonly
m=audio 20000 RTP/AVP 0 8 97
i=Substitutive audio RTP Stream
c=IN IP4 substitive.example.com
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=sendonly
a=mid:1
m=video 20002 RTP/AVP 31 32
i=Substitutive video RTP Stream
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c=IN IP4 substitive.example.com
a=rtpmap:31 H261/90000
a=rtpmap:32 MPV/90000
a=mid:2
a=sendonly
SDP Answer - from the splicer
v=0
o=bob 2808844564 2808844564 IN IP4 splicer.example.com
s=RTP Splicing Example
c=IN IP4 splicer.example.com
t=0 0
a=group:SPLICE foo 1
a=group:SPLICE bar 2
a=group:BUNDLE foo bar
m=audio 30000 RTP/AVP 0
a=mid:foo
b=AS:200
a=rtpmap:0 PCMU/8000
a=extmap:1 urn:ietf:params:rtp-hdrext:splicing-interval
a=recvonly
m=video 30000 RTP/AVP 32
a=mid:bar
b=AS:1000
a=rtpmap:32 MPV/90000
a=extmap:2 urn:ietf:params:rtp-hdrext:splicing-interval
a=recvonly
m=audio 30002 RTP/AVP 0
i=Substitutive audio RTP Stream
c=IN IP4 splicer.example.com
a=rtpmap:0 PCMU/8000
a=recvonly
a=mid:1
m=video 30004 RTP/AVP 32
i=Substitutive video RTP Stream
c=IN IP4 splicer.example.com
a=rtpmap:32 MPV/90000
a=mid:2
a=recvonly
6.4 Offer/Answer with BUNDLE: a Subset of Media are Spliced
In this example, the substitutive media only applies for video when
splicing:
1. An Offer, in which the offerer assigns a unique address to each
bundled "m="line within the BUNDLE group, and assigns a substitutive
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media to the bundled video "m=" line for splicing.
2. An answer, in which the answerer selects its own BUNDLE address,
and leave the substitutive media untouched.
SDP Offer - from the main RTP sender:
v=0
o=alice 1122334455 1122334466 IN IP4 splicing.example.com
s=RTP Splicing Example
c=IN IP4 splicing.example.com
t=0 0
a=group:SPLICE bar 2
a=group:BUNDLE foo bar
m=audio 10000 RTP/AVP 0 8 97
a=mid:foo
b=AS:200
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=sendonly
m=video 10002 RTP/AVP 31 32
a=mid:bar
b=AS:1000
a=rtpmap:31 H261/90000
a=rtpmap:32 MPV/90000
a=extmap:2 urn:ietf:params:rtp-hdrext:splicing-interval
a=sendonly
m=video 20000 RTP/AVP 31 32
i=Substitutive video RTP Stream
c=IN IP4 substitutive.example.com
a=rtpmap:31 H261/90000
a=rtpmap:32 MPV/90000
a=mid:2
a=sendonly
SDP Answer - from the splicer:
v=0
o=bob 2808844564 2808844564 IN IP4 splicer.example.com
s=RTP Splicing Example
c=IN IP4 splicer.example.com
t=0 0
a=group:SPLICE bar 2
a=group:BUNDLE foo bar
m=audio 30000 RTP/AVP 0
a=mid:foo
b=AS:200
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a=rtpmap:0 PCMU/8000
a=recvonly
m=video 30000 RTP/AVP 32
a=mid:bar
b=AS:1000
a=rtpmap:32 MPV/90000
a=extmap:2 urn:ietf:params:rtp-hdrext:splicing-interval
a=recvonly
m=video 30004 RTP/AVP 32
i=Substitutive video RTP Stream
c=IN IP4 splicer.example.com
a=rtpmap:32 MPV/90000
a=mid:2
a=recvonly
7 Security Considerations
The security considerations of the RTP specification [RFC3550] and
the general mechanism for RTP header extensions [RFC5285bis] apply.
The splicer can either be a mixer or a translator, and all the
security considerations of these two RTP intermediaries topologies
described in [RFC7667] and [RFC7201] are applicable for the splicer.
The splicer replaces some content with other content in RTP packet,
thus breaking any RTP-level end-to-end security, such as source
authentication and integrity protection. End to end source
authentication is not possible with any known existing splicing
solution. A new solution can theoretically be developed that enables
identification of the participating entities and what each provides,
i.e., the different media sources, main and substituting, and the
splicer which provides the RTP-level integration of the media
payloads in a common timeline and synchronization context.
Since the splicer breaks RTP-level end-to-end security, it needs to
be part of the signaling context and the necessary security
associations (e.g., SRTP crypto contexts) established for the RTP
session participants. When using the Secure Real-Time Transport
Protocol (SRTP) [RFC3711], the splicer would have to be provisioned
with the same security association as the main RTP sender.
If there is a concern about the confidentiality of the splicing time
information, the header extension defined in this document MUST be
also protected, for example, header extension encryption [RFC6904]
can be used in this case. However, the malicious endpoint may get the
splicing time information by other means, e.g., inferring from the
communication between the main and substitutive content sources. To
avoid the insertion of invalid substitutive content, the splicer MUST
have some mechanisms to authenticate the substitutive stream source.
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For cases that the splicing time information is changed by a
malicious endpoint, the splicing, for example, may fail since it will
not be available at the right time for the substitutive media to
arrive. Another case is that an attacker may prevent the receivers
receiving the content from the main sender by inserting extra
splicing time information. To avoid the above cases happening, the
authentication of the RTP header extension for splicing time
information SHOULD be considered.
When a splicer implemented as a mixer sends the stream to the
receivers, CSRC list, which can be used to detect RTP-level
forwarding loops as defined in Section 8.2 of [RFC3550], may be
removed for simplifying the receivers that can not handle multiple
sources in the RTP stream. Hence, loops may occur to cause packets to
loop back to upstream of the splicer and may form a serious denial-
of-service threat. In such a case, non-RTP means, e.g., signaling
among all the participants, MUST be used to detect and resolve loops.
8 IANA Considerations
8.1 RTCP Control Packet Types
Based on the guidelines suggested in [RFC5226], a new RTCP packet
format has been registered with the RTCP Control Packet Type (PT)
Registry:
Name: SNM
Long name: Splicing Notification Message
Value: TBA
Reference: This document
8.2 RTP Compact Header Extensions
The IANA has also registered a new RTP Compact Header Extension
[RFC5285bis], according to the following:
Extension URI: urn:ietf:params:rtp-hdrext:splicing-interval
Description: Splicing Interval
Contact: Jinwei Xia <xiajinwei@huawei.com>
Reference: This document
8.3 SDP Grouping Semantic Extension
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This document request IANA to register the new SDP grouping semantic
extension called "SPLICE".
Semantics: Splice
Token:SPLICE
Reference: This document
9 Acknowledgement
The authors would like to thank the following individuals who help to
review this document and provide very valuable comments: Colin
Perkins, Bo Burman, Stephen Botzko, Ben Campbell.
10 References
10.1 Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3264] Rosenberg, J., and H. Schulzrinne, "An Offer/Answer Model
with the Session Description Protocol (SDP)", RFC 3264,
June 2002.
[RFC5285bis] Even, R., Singer, D. and H. Desineni, "A General
Mechanism for RTP Header Extensions", draft-ietf-avtcore-
rfc5285-bis-02, May 2016.
[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description
Protocol (SDP) Grouping Framework", RFC 5888, June 2010.
[RFC5905] Mills, D., Martin, J., Ed., Burbank, J., and W. Kasch,
"Network Time Protocol Version 4: Protocol and Algorithms
Specification", RFC 5905, June 2010.
[RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
Flows", RFC 6051, November 2010.
[RFC7201] Westerlund, M. and C. Perkins, "Options for Securing RTP
Sessions", RFC 7201, April 2014.
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[RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
November 2015.
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.
[RFC5226] Narten, T. and H. Alvestrand, "Guidelines for Writing an
IANA Considerations Section in RFCs", BCP 26, RFC 5226,
May 2008.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009.
[RFC6285] Ver Steeg, B., Begen, A., Van Caenegem, T., and Z. Vax,
"Unicast-Based Rapid Acquisition of Multicast RTP
Sessions", RFC 6285, June 2011.
[RFC6904] Lennox, J.,"Encryption of Header Extensions in the Secure
Real-Time Transport Protocol (SRTP)", April 2013.
[SCTE35] Society of Cable Telecommunications Engineers (SCTE),
"Digital Program Insertion Cueing Message for Cable",
2011.
[RFC6828] Xia, J., "Content Splicing for RTP Sessions", RFC 6828,
January 2013.
Authors' Addresses
Jinwei Xia
Huawei
Email: xiajinwei@huawei.com
Roni Even
Huawei
Email: ron.even.tlv@gmail.com
Rachel Huang
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Huawei
Email: rachel.huang@huawei.com
Lingli Deng
China Mobile
Email: denglingli@chinamobile.com
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