RTP Payload for Timed Text Markup Language (TTML)British Broadcasting CorporationDock House, MediaCityUKSalfordUnited Kingdom+44 30304 09549james.sandford@bbc.co.uk
Internet
Audio/Video Transport Core Maintenance Working GroupsubtitlescaptionsimscmediastreamingsdpxmlThis memo describes a Real-time Transport Protocol (RTP) payload format for
Timed Text Markup Language (TTML), an XML-based timed text format from
W3C. This payload format is specifically targeted at streaming workflows using
TTML.Status of This Memo
This is an Internet Standards Track document.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by
the Internet Engineering Steering Group (IESG). Further
information on Internet Standards is available in Section 2 of
RFC 7841.
Information about the current status of this document, any
errata, and how to provide feedback on it may be obtained at
.
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Table of Contents
IntroductionTTML (Timed Text Markup Language) is a media type for
describing timed text, such as closed captions and subtitles in television
workflows or broadcasts, as XML. This document specifies how TTML should be
mapped into an RTP stream in streaming workflows, including (but not restricted
to) those described in the television-broadcast-oriented European Broadcasting
Union Timed Text (EBU-TT) Part 3 specification. This document does not define a media type
for TTML but makes use of the existing application/ttml+xml media type .Conventions and DefinitionsUnless otherwise stated, the term "document" refers to the TTML document
being transmitted in the payload of the RTP packet(s).The term "word" refers to a data word aligned to a specified number of bits
in a computing sense and not to linguistic words that might appear in
the transported text.
The key words "MUST", "MUST NOT",
"REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT",
"RECOMMENDED", "NOT RECOMMENDED",
"MAY", and "OPTIONAL" in this document are
to be interpreted as
described in BCP 14
when, and only when, they appear in all capitals, as shown here.
Media Format DescriptionRelation to Other Text Payload TypesPrior payload types for text are not suited to the carriage of closed
captions in television workflows. "RTP Payload for Text Conversation" is intended for low data rate conversation with its own
session management and minimal formatting capabilities. "Definition of Events for
Modem, Fax, and Text Telephony Signals" deals in large
parts with the control signalling of facsimile and other systems. "RTP Payload Format for
3rd Generation Partnership Project (3GPP) Timed Text"
describes the carriage of a timed text format with much more restricted
formatting capabilities than TTML. The lack of an existing format for TTML or
generic XML has necessitated the creation of this payload format.TTML2TTML2 (Timed Text Markup Language, Version 2) is an
XML-based markup language for describing textual information with associated
timing metadata. One of its primary use cases is the description of subtitles
and closed captions. A number of profiles exist that adapt TTML2 for use in
specific contexts . These include both file-based
and streaming workflows.Payload FormatIn addition to the required RTP headers, the payload contains a section for
the TTML document being transmitted (User Data Words) and a field for the
length of that data. Each RTP payload contains one or part of one TTML
document.A representation of the payload format for TTML is .RTP Header UsageRTP packet header fields SHALL be interpreted, as per , with the following specifics:
Marker Bit (M): 1 bit
The marker bit is set to "1" to indicate the last packet of a
document. Otherwise, set to "0". Note: The first packet might also be the
last.
Timestamp: 32 bits
The RTP Timestamp encodes the epoch of the TTML document in User Data
Words. Further detail on its usage may be found in . The clock frequency used is dependent on the
application and is specified in the media type rate parameter, as per . Documents spread across multiple packets MUST
use the same timestamp but different consecutive Sequence Numbers. Sequential
documents MUST NOT use the same timestamp. Because packets do
not represent any constant duration, the timestamp cannot be used to directly
infer packet loss.
Reserved: 16 bits
These bits are reserved for future use and MUST be set to
0x0 and ignored upon reception.
Length: 16 bits
The length of User Data Words in bytes.
User Data Words: The length of User Data Words MUST match
the value specified in the Length field
The User Data Words section contains the text of the whole document being transmitted
or a part of the document being transmitted. Documents using character
encodings where characters are not represented by a single byte
MUST be serialised in big-endian order, a.k.a., network byte
order. Where a document will not fit within the Path MTU, it may be fragmented
across multiple packets. Further detail on fragmentation may be found in .
Payload DataTTML documents define a series of changes to text over time. TTML documents
carried in User Data Words are encoded in accordance with one or more of the
defined TTML profiles specified in the TTML registry . These profiles specify the document structure used,
systems models, timing, and other considerations. TTML profiles may restrict
the complexity of the changes, and operational requirements may limit the
maximum duration of TTML documents by a deployment configuration. Both of
these cases are out of scope of this document.Documents carried over RTP MUST conform to the following
profile, in addition to any others used.Payload Content RestrictionsThis section defines constraints on the content of TTML documents carried
over RTP.Multiple TTML subtitle streams MUST NOT be interleaved in a
single RTP stream.The TTML document instance's root tt element in the
http://www.w3.org/ns/ttml namespace MUST include a
timeBase attribute in the
http://www.w3.org/ns/ttml#parameter namespace containing the value
media.This is equivalent to the TTML2 content profile definition document in
.Payload Processing RequirementsThis section defines constraints on the processing of the TTML documents carried over RTP.If a TTML document is assessed to be invalid, then it MUST be
discarded. This includes empty documents, i.e., those of zero length. When
processing a valid document, the following requirements apply.Each TTML document becomes active at its epoch E. E MUST be
set to the RTP Timestamp in the header of the RTP packet carrying the TTML
document. Computed TTML media times are offset relative to E, in accordance
with Section I.2 of .When processing a sequence of TTML documents, where each is delivered in
the same RTP stream, exactly zero or one document SHALL be
considered active at each moment in the RTP time line.
In the event that a document
Dn-1 with En-1 is active, and document Dn is
delivered with En where En-1 < En,
processing of Dn-1MUST be stopped at En
and processing of DnMUST begin.When all defined content within a document has ended, then processing of the
document MAY be stopped. This can be tested by constructing the
intermediate synchronic document sequence from the document, as defined by
. If the last intermediate synchronic document in the
sequence is both active and contains no region elements, then all defined
content within the document has ended.As described above, the RTP Timestamp does not specify the exact timing of
the media in this payload format. Additionally, documents may be fragmented
across multiple packets. This renders the RTCP jitter calculation
unusable.TTML Processor ProfileFeature Extension DesignationThis specification defines the following TTML feature extension designation:
urn:ietf:rfc:8759#rtp-relative-media-time
The namespace urn:ietf:rfc:8759 is as defined by .A TTML content processor supports the #rtp-relative-media-time
feature extension if it processes media times in accordance with the payload
processing requirements specified in this document, i.e., that the epoch E is
set to the time equivalent to the RTP Timestamp, as detailed above in .Processor Profile DocumentThe required syntax and semantics declared in the minimal TTML2 processor
profile in MUST be supported by
the receiver,
as signified by those feature or extension elements whose
value attribute is set to required.Note that this requirement does not imply that the receiver needs to
support either TTML1 or TTML2 profile processing, i.e., the TTML2
#profile-full-version-2 feature or any of
its dependent features.Processor Profile SignallingThe codecs media type parameter MUST specify at
least one processor profile. Short codes for TTML profiles are registered at
. The processor profiles specified in
codecsMUST be compatible with the processor profile
specified in this document. Where multiple options exist in codecs
for possible processor profile combinations (i.e., separated by |
operator), every permitted option MUST be compatible with the
processor profile specified in this document. Where processor profiles (other
than the one specified in this document) are advertised in the codecs
parameter, the requirements of the processor profile specified in this
document MAY be signalled, additionally using the +
operator with its registered short code.A processor profile (X) is compatible with the processor profile specified
here (P) if X includes all the features and extensions in P (identified by
their character content) and the value attribute of each is, at least,
as restrictive as the value attribute of the feature or extension in
P that has the same character content. The term "restrictive" here is as
defined in Section 6 of .Payload Examples is an example of a valid TTML document that may
be carried using the payload format described in this document.Fragmentation of TTML DocumentsMany of the use cases for TTML are low bit-rate with RTP packets expected
to fit within the Path MTU. However, some documents may exceed the Path
MTU. In these cases, they may be split between multiple packets. Where
fragmentation is used, the following guidelines MUST be
followed:
It is RECOMMENDED that documents be fragmented as seldom
as possible, i.e., the least possible number of fragments is created out of a
document.
Text strings MUST split at character boundaries. This
enables decoding of partial documents. As a consequence, document
fragmentation requires knowledge of the UTF-8/UTF-16 encoding formats to
determine character boundaries.
Document fragments SHOULD be protected against packet
losses. More information can be found in .
When a document spans more than one RTP packet, the entire document is
obtained by concatenating User Data Words from each consecutive contributing
packet in ascending order of Sequence Number.As described in , only zero or one TTML
document may be active at any point in time. As such, there
MUST only be one document transmitted for a given RTP
Timestamp. Furthermore, as stated in , the
marker bit MUST be set for a packet containing the last
fragment of a document. A packet following one where the marker bit is set
contains the first fragment of a new document. The first fragment might also
be the last.Protection against Loss of DataConsideration must be devoted to keeping loss of documents due to packet
loss within acceptable limits. What is deemed acceptable limits is dependent
on the TTML profile(s) used and use case, among other things. As such, specific
limits are outside the scope of this document.Documents MAY be sent without additional protection if
end-to-end network conditions guarantee that document loss will be within
acceptable limits under all anticipated load conditions. Where such guarantees
cannot be provided, implementations MUST use a mechanism to
protect against packet loss. Potential mechanisms include Forward Error
Correction (FEC) , retransmission , duplication , or an equivalent
technique.Congestion Control ConsiderationsCongestion control for RTP SHALL be used in accordance with
and with any applicable RTP profile, e.g., . "Multimedia Congestion Control: Circuit Breakers for
Unicast RTP Sessions" is an update to
"RTP: A Transport Protocol for Real-time
Applications" , which defines criteria for when one is required to
stop sending RTP packet streams. Applications implementing this standard
MUST comply with , with particular
attention paid to Section
on Media Usability. provides additional information
on the best practices for applying congestion control to UDP streams.Payload Format ParametersThis RTP payload format is identified using the existing
application/ttml+xml media type as registered with IANA
and defined in .Clock RateThe default clock rate for TTML over RTP is 1000 Hz. The clock rate
SHOULD be included in any advertisements of the RTP stream
where possible. This parameter has not been added to the media type definition
as it is not applicable to TTML usage other than within RTP streams. In other
contexts, timing is defined within the TTML document.When choosing a clock rate, implementers should consider what other media
their TTML streams may be used in conjunction with (e.g., video or audio). In
these situations, it is RECOMMENDED that streams use the same
clock source and clock rate as the related media. As TTML streams may be
aperiodic, implementers should also consider the frequency range over which
they expect packets to be sent and the temporal resolution required.Session Description Protocol (SDP) ConsiderationsThe mapping of the application/ttml+xml media type and its parameters SHALL be done according to
.
The type name "application" goes in SDP "m=" as the media name.
The media subtype "ttml+xml" goes in SDP "a=rtpmap" as the encoding name.
The clock rate also goes in "a=rtpmap" as the clock rate.
Additional format-specific parameters, as described in the media type
specification, SHALL be included in the SDP file in "a=fmtp" as
a semicolon-separated list of "parameter=value" pairs, as described in . The codecs parameter MUST be
included in the a=fmtp line of the SDP file. Specific requirements
for the "codecs" parameter are included in .ExamplesA sample SDP mapping is presented in .In this example, a dynamic payload type 112 is used. The 90 kHz RTP
timestamp rate is specified in the "a=rtpmap" line after the subtype.
The codecs parameter defined in the "a=fmtp" line indicates that the TTML data
conforms to Internet Media and Captions (IMSC) 1.1 Text profile .IANA ConsiderationsThis document has no IANA actions.Security ConsiderationsRTP packets using the payload format defined in this specification are
subject to the security considerations discussed in the RTP specification
and in any applicable RTP profile, such as RTP/AVP
, RTP/AVPF , RTP/SAVP , or RTP/SAVPF .
However, as
"Securing the RTP Protocol Framework: Why RTP Does Not Mandate a Single Media
Security Solution" discusses, it is not an RTP
payload format's responsibility to discuss or mandate what solutions are used
to meet the basic security goals (like confidentiality, integrity, and source
authenticity) for RTP in general. This responsibility lays on anyone using RTP
in an application. They can find guidance on available security mechanisms
and important considerations in "Options for Securing RTP Sessions" . Applications SHOULD use one or more
appropriate strong security mechanisms. The rest of this Security
Considerations section discusses the security impacting properties of the
payload format itself.To avoid potential buffer overflow attacks, receivers should take care to
validate that the User Data Words in the RTP payload are of the appropriate
length (using the Length field).This payload format places no specific restrictions on the size of TTML
documents that may be transmitted. As such, malicious implementations could be
used to perform denial-of-service (DoS) attacks. provides more information on DoS attacks and describes some
mitigation strategies. Implementers should take into consideration that the
size and frequency of documents transmitted using this format may vary over
time. As such, sender implementations should avoid producing streams that
exhibit DoS-like behaviour, and receivers should avoid false identification of
a legitimate stream as malicious.As with other XML types and as noted in "XML Media Types",
repeated expansion of maliciously constructed XML
entities can be used to consume large amounts of memory, which may cause XML
processors in constrained environments to fail.In addition, because of the extensibility features for TTML and of XML in
general, it is possible that "application/ttml+xml" may describe content that
has security implications beyond those described here. However, TTML does not
provide for any sort of active or executable content, and if the processor
follows only the normative semantics of the published specification, this
content will be outside TTML namespaces and may be ignored. Only in the case
where the processor recognizes and processes the additional content or where
further processing of that content is dispatched to other processors would
security issues potentially arise. And in that case, they would fall outside
the domain of this RTP payload format and the application/ttml+xml
registration document.Although not prohibited, there are no expectations that XML signatures or
encryption would normally be employed.Further information related to privacy and security at a document level can
be found in Appendix P of .Normative ReferencesMedia TypesIANAKey words for use in RFCs to Indicate Requirement LevelsIn many standards track documents several words are used to signify the requirements in the specification. These words are often capitalized. This document defines these words as they should be interpreted in IETF documents. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.RTP: A Transport Protocol for Real-Time ApplicationsThis memorandum describes RTP, the real-time transport protocol. RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services. RTP does not address resource reservation and does not guarantee quality-of- service for real-time services. The data transport is augmented by a control protocol (RTCP) to allow monitoring of the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality. RTP and RTCP are designed to be independent of the underlying transport and network layers. The protocol supports the use of RTP-level translators and mixers. Most of the text in this memorandum is identical to RFC 1889 which it obsoletes. There are no changes in the packet formats on the wire, only changes to the rules and algorithms governing how the protocol is used. The biggest change is an enhancement to the scalable timer algorithm for calculating when to send RTCP packets in order to minimize transmission in excess of the intended rate when many participants join a session simultaneously. [STANDARDS-TRACK]RTP Payload for Text ConversationThis memo obsoletes RFC 2793; it describes how to carry real-time text conversation session contents in RTP packets. Text conversation session contents are specified in ITU-T Recommendation T.140.One payload format is described for transmitting text on a separate RTP session dedicated for the transmission of text.This RTP payload description recommends a method to include redundant text from already transmitted packets in order to reduce the risk of text loss caused by packet loss. [STANDARDS-TRACK]Media Type Registration of RTP Payload FormatsThis document specifies the procedure to register RTP payload formats as audio, video, or other media subtype names. This is useful in a text-based format description or control protocol to identify the type of an RTP transmission. [STANDARDS-TRACK]XML Media TypesThis specification standardizes three media types -- application/xml, application/xml-external-parsed-entity, and application/xml-dtd -- for use in exchanging network entities that are related to the Extensible Markup Language (XML) while defining text/xml and text/ xml-external-parsed-entity as aliases for the respective application/ types. This specification also standardizes the '+xml' suffix for naming media types outside of these five types when those media types represent XML MIME entities.Multimedia Congestion Control: Circuit Breakers for Unicast RTP SessionsThe Real-time Transport Protocol (RTP) is widely used in telephony, video conferencing, and telepresence applications. Such applications are often run on best-effort UDP/IP networks. If congestion control is not implemented in these applications, then network congestion can lead to uncontrolled packet loss and a resulting deterioration of the user's multimedia experience. The congestion control algorithm acts as a safety measure by stopping RTP flows from using excessive resources and protecting the network from overload. At the time of this writing, however, while there are several proprietary solutions, there is no standard algorithm for congestion control of interactive RTP flows.This document does not propose a congestion control algorithm. It instead defines a minimal set of RTP circuit breakers: conditions under which an RTP sender needs to stop transmitting media data to protect the network from excessive congestion. It is expected that, in the absence of long-lived excessive congestion, RTP applications running on best-effort IP networks will be able to operate without triggering these circuit breakers. To avoid triggering the RTP circuit breaker, any Standards Track congestion control algorithms defined for RTP will need to operate within the envelope set by these RTP circuit breaker algorithms.UDP Usage GuidelinesThe User Datagram Protocol (UDP) provides a minimal message-passing transport that has no inherent congestion control mechanisms. This document provides guidelines on the use of UDP for the designers of applications, tunnels, and other protocols that use UDP. Congestion control guidelines are a primary focus, but the document also provides guidance on other topics, including message sizes, reliability, checksums, middlebox traversal, the use of Explicit Congestion Notification (ECN), Differentiated Services Code Points (DSCPs), and ports.Because congestion control is critical to the stable operation of the Internet, applications and other protocols that choose to use UDP as an Internet transport must employ mechanisms to prevent congestion collapse and to establish some degree of fairness with concurrent traffic. They may also need to implement additional mechanisms, depending on how they use UDP.Some guidance is also applicable to the design of other protocols (e.g., protocols layered directly on IP or via IP-based tunnels), especially when these protocols do not themselves provide congestion control.This document obsoletes RFC 5405 and adds guidelines for multicast UDP usage.Ambiguity of Uppercase vs Lowercase in RFC 2119 Key WordsRFC 2119 specifies common key words that may be used in protocol specifications. This document aims to reduce the ambiguity by clarifying that only UPPERCASE usage of the key words have the defined special meanings.EBU-TT, Part 3, Live Subtitling Applications: System Model and Content Profile for Authoring and ContributionEuropean Broadcasting UnionTTML Media Type Definition and Profile RegistryW3C Working Group NoteTimed Text Markup Language 2 (TTML2)Informative ReferencesA URN Namespace for IETF DocumentsThis document proposes the "ietf" namespace, which consists of the RFC family of documents (RFCs, STDs, FYIs, and BCPs) developed by the IETF and published by the RFC Editor and the minutes of working groups (WG) and birds of a feather (BOF) meetings that occur during IETF conferences. [STANDARDS-TRACK]RTP Profile for Audio and Video Conferences with Minimal ControlThis document describes a profile called "RTP/AVP" for the use of the real-time transport protocol (RTP), version 2, and the associated control protocol, RTCP, within audio and video multiparticipant conferences with minimal control. It provides interpretations of generic fields within the RTP specification suitable for audio and video conferences. In particular, this document defines a set of default mappings from payload type numbers to encodings. This document also describes how audio and video data may be carried within RTP. It defines a set of standard encodings and their names when used within RTP. The descriptions provide pointers to reference implementations and the detailed standards. This document is meant as an aid for implementors of audio, video and other real-time multimedia applications. This memorandum obsoletes RFC 1890. It is mostly backwards-compatible except for functions removed because two interoperable implementations were not found. The additions to RFC 1890 codify existing practice in the use of payload formats under this profile and include new payload formats defined since RFC 1890 was published. [STANDARDS-TRACK]The Secure Real-time Transport Protocol (SRTP)This document describes the Secure Real-time Transport Protocol (SRTP), a profile of the Real-time Transport Protocol (RTP), which can provide confidentiality, message authentication, and replay protection to the RTP traffic and to the control traffic for RTP, the Real-time Transport Control Protocol (RTCP). [STANDARDS-TRACK]RTP Payload Format for 3rd Generation Partnership Project (3GPP) Timed TextThis document specifies an RTP payload format for the transmission of 3GPP (3rd Generation Partnership Project) timed text. 3GPP timed text is a time-lined, decorated text media format with defined storage in a 3GP file. Timed Text can be synchronized with audio/video contents and used in applications such as captioning, titling, and multimedia presentations. In the following sections, the problems of streaming timed text are addressed, and a payload format for streaming 3GPP timed text over RTP is specified. [STANDARDS-TRACK]Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)Real-time media streams that use RTP are, to some degree, resilient against packet losses. Receivers may use the base mechanisms of the Real-time Transport Control Protocol (RTCP) to report packet reception statistics and thus allow a sender to adapt its transmission behavior in the mid-term. This is the sole means for feedback and feedback-based error repair (besides a few codec-specific mechanisms). This document defines an extension to the Audio-visual Profile (AVP) that enables receivers to provide, statistically, more immediate feedback to the senders and thus allows for short-term adaptation and efficient feedback-based repair mechanisms to be implemented. This early feedback profile (AVPF) maintains the AVP bandwidth constraints for RTCP and preserves scalability to large groups. [STANDARDS-TRACK]RTP Retransmission Payload FormatRTP retransmission is an effective packet loss recovery technique for real-time applications with relaxed delay bounds. This document describes an RTP payload format for performing retransmissions. Retransmitted RTP packets are sent in a separate stream from the original RTP stream. It is assumed that feedback from receivers to senders is available. In particular, it is assumed that Real-time Transport Control Protocol (RTCP) feedback as defined in the extended RTP profile for RTCP-based feedback (denoted RTP/AVPF) is available in this memo. [STANDARDS-TRACK]Internet Denial-of-Service ConsiderationsIABThis document provides an overview of possible avenues for denial-of-service (DoS) attack on Internet systems. The aim is to encourage protocol designers and network engineers towards designs that are more robust. We discuss partial solutions that reduce the effectiveness of attacks, and how some solutions might inadvertently open up alternative vulnerabilities. This memo provides information for the Internet community.Definition of Events for Modem, Fax, and Text Telephony SignalsThis memo updates RFC 4733 to add event codes for modem, fax, and text telephony signals when carried in the telephony event RTP payload. It supersedes the assignment of event codes for this purpose in RFC 2833, and therefore obsoletes that part of RFC 2833. [STANDARDS-TRACK]RTP Payload Format for Generic Forward Error CorrectionThis document specifies a payload format for generic Forward Error Correction (FEC) for media data encapsulated in RTP. It is based on the exclusive-or (parity) operation. The payload format described in this document allows end systems to apply protection using various protection lengths and levels, in addition to using various protection group sizes to adapt to different media and channel characteristics. It enables complete recovery of the protected packets or partial recovery of the critical parts of the payload depending on the packet loss situation. This scheme is completely compatible with non-FEC-capable hosts, so the receivers in a multicast group that do not implement FEC can still work by simply ignoring the protection data. This specification obsoletes RFC 2733 and RFC 3009. The FEC specified in this document is not backward compatible with RFC 2733 and RFC 3009. [STANDARDS-TRACK]Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)An RTP profile (SAVP) for secure real-time communications and another profile (AVPF) to provide timely feedback from the receivers to a sender are defined in RFC 3711 and RFC 4585, respectively. This memo specifies the combination of both profiles to enable secure RTP communications with feedback. [STANDARDS-TRACK]Options for Securing RTP SessionsThe Real-time Transport Protocol (RTP) is used in a large number of different application domains and environments. This heterogeneity implies that different security mechanisms are needed to provide services such as confidentiality, integrity, and source authentication of RTP and RTP Control Protocol (RTCP) packets suitable for the various environments. The range of solutions makes it difficult for RTP-based application developers to pick the most suitable mechanism. This document provides an overview of a number of security solutions for RTP and gives guidance for developers on how to choose the appropriate security mechanism.Securing the RTP Framework: Why RTP Does Not Mandate a Single Media Security SolutionThis memo discusses the problem of securing real-time multimedia sessions. It also explains why the Real-time Transport Protocol (RTP) and the associated RTP Control Protocol (RTCP) do not mandate a single media security mechanism. This is relevant for designers and reviewers of future RTP extensions to ensure that appropriate security mechanisms are mandated and that any such mechanisms are specified in a manner that conforms with the RTP architecture.Seamless Protection Switching of RTP DatagramsSMPTETTML Profiles for Internet Media Subtitles and Captions 1.1AcknowledgementsThanks to , , , , , , , and for their valuable
feedback throughout the
development of this document. Thanks to the W3C Timed Text Working Group and
EBU Timed Text Working Group for their substantial efforts in developing the
timed text format this payload format is intended to carry.Author's AddressBritish Broadcasting CorporationDock House, MediaCityUKSalfordUnited Kingdom+44 30304 09549james.sandford@bbc.co.uk