AVT B. VerSteeg
Internet-Draft A. Begen
Intended status: Standards Track Cisco
Expires: January 10, 2011 T. VanCaenegem
Alcatel-Lucent
Z. Vax
Microsoft Corporation
July 9, 2010
Unicast-Based Rapid Acquisition of Multicast RTP Sessions
draft-ietf-avt-rapid-acquisition-for-rtp-11
Abstract
When an RTP receiver joins a multicast session, it may need to
acquire and parse certain Reference Information before it can process
any data sent in the multicast session. Depending on the join time,
length of the Reference Information repetition (or appearance)
interval, size of the Reference Information as well as the
application and transport properties, the time lag before an RTP
receiver can usefully consume the multicast data, which we refer to
as the Acquisition Delay, varies and can be large. This is an
undesirable phenomenon for receivers that frequently switch among
different multicast sessions, such as video broadcasts.
In this document, we describe a method using the existing RTP and
RTCP protocol machinery that reduces the acquisition delay. In this
method, an auxiliary unicast RTP session carrying the Reference
Information to the receiver precedes/accompanies the multicast
stream. This unicast RTP flow can be transmitted at a faster than
natural bitrate to further accelerate the acquisition. The
motivating use case for this capability is multicast applications
that carry real-time compressed audio and video. However, the
proposed method can also be used in other types of multicast
applications where the acquisition delay is long enough to be a
problem.
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
1.1. Acquisition of RTP Streams vs. RTP Sessions . . . . . . . 6
1.2. Outline . . . . . . . . . . . . . . . . . . . . . . . . . 7
2. Requirements Notation . . . . . . . . . . . . . . . . . . . . 7
3. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 7
4. Elements of Delay in Multicast Applications . . . . . . . . . 9
5. Protocol Design Considerations and Their Effect on
Resource Management for Rapid Acquisition . . . . . . . . . . 10
6. Rapid Acquisition of Multicast RTP Sessions (RAMS) . . . . . . 12
6.1. Overview . . . . . . . . . . . . . . . . . . . . . . . . . 12
6.2. Message Flows . . . . . . . . . . . . . . . . . . . . . . 13
6.3. Synchronization of Primary Multicast Streams . . . . . . . 23
6.4. Burst Shaping and Congestion Control in RAMS . . . . . . . 23
6.5. Failure Cases . . . . . . . . . . . . . . . . . . . . . . 26
7. Encoding of the Signaling Protocol in RTCP . . . . . . . . . . 27
7.1. Extensions . . . . . . . . . . . . . . . . . . . . . . . . 28
7.1.1. Vendor-Neutral Extensions . . . . . . . . . . . . . . 29
7.1.2. Private Extensions . . . . . . . . . . . . . . . . . . 29
7.2. RAMS Request . . . . . . . . . . . . . . . . . . . . . . . 29
7.3. RAMS Information . . . . . . . . . . . . . . . . . . . . . 32
7.4. RAMS Termination . . . . . . . . . . . . . . . . . . . . . 34
8. SDP Signaling . . . . . . . . . . . . . . . . . . . . . . . . 35
8.1. Definitions . . . . . . . . . . . . . . . . . . . . . . . 35
8.2. Requirements . . . . . . . . . . . . . . . . . . . . . . . 36
8.3. Example and Discussion . . . . . . . . . . . . . . . . . . 37
9. NAT Considerations . . . . . . . . . . . . . . . . . . . . . . 39
10. Security Considerations . . . . . . . . . . . . . . . . . . . 40
11. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 42
11.1. Registration of SDP Attributes . . . . . . . . . . . . . . 42
11.2. Registration of SDP Attribute Values . . . . . . . . . . . 42
11.3. Registration of FMT Values . . . . . . . . . . . . . . . . 43
11.4. SFMT Values for RAMS Messages Registry . . . . . . . . . . 43
11.5. RAMS TLV Space Registry . . . . . . . . . . . . . . . . . 44
11.6. RAMS Response Code Space Registry . . . . . . . . . . . . 45
12. Contributors . . . . . . . . . . . . . . . . . . . . . . . . . 47
13. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 47
14. References . . . . . . . . . . . . . . . . . . . . . . . . . . 47
14.1. Normative References . . . . . . . . . . . . . . . . . . . 47
14.2. Informative References . . . . . . . . . . . . . . . . . . 49
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 50
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1. Introduction
Most multicast flows carry a stream of inter-related data. The
receivers need to acquire certain information to start processing any
data sent in the multicast session. This document refers to this
information as Reference Information. The Reference Information is
conventionally sent periodically in the multicast session (although
its content can change over time) and usually consists of items such
as a description of the schema for the rest of the data, references
to which data to process, encryption information including keys, as
well as any other information required to process the data in the
multicast stream [IC2009].
Real-time multicast applications require the receivers to buffer
data. The receiver may have to buffer data to smooth out the network
jitter, to allow loss-repair methods such as Forward Error Correction
and retransmission to recover the missing packets, and to satisfy the
data processing requirements of the application layer.
When a receiver joins a multicast session, it has no control over
what point in the flow is currently being transmitted. Sometimes the
receiver might join the session right before the Reference
Information is sent in the session. In this case, the required
waiting time is usually minimal. Other times, the receiver might
join the session right after the Reference Information has been
transmitted. In this case, the receiver has to wait for the
Reference Information to appear again in the flow before it can start
processing any multicast data. In some other cases, the Reference
Information is not contiguous in the flow but dispersed over a large
period, which forces the receiver to wait for all of the Reference
Information to arrive before starting to process the rest of the
data.
The net effect of waiting for the Reference Information and waiting
for various buffers to fill up is that the receivers can experience
significantly large delays in data processing. In this document, we
refer to the difference between the time an RTP receiver joins the
multicast session and the time the RTP receiver acquires all the
necessary Reference Information as the Acquisition Delay. The
acquisition delay might not be the same for different receivers; it
usually varies depending on the join time, length of the Reference
Information repetition (or appearance) interval, size of the
Reference Information as well as the application and transport
properties.
The varying nature of the acquisition delay adversely affects the
receivers that frequently switch among multicast sessions. In this
specification, we address this problem for RTP-based multicast
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applications and describe a method that uses the fundamental tools
offered by the existing RTP and RTCP protocols [RFC3550]. In this
method, either the multicast source (or the distribution source in a
source-specific multicast (SSM) session) retains the Reference
Information for a period after its transmission, or an intermediary
network element (that we refer to as Retransmission Server) joins the
multicast session and continuously caches the Reference Information
as it is sent in the session and acts as a feedback target (See
[RFC5760]) for the session. When an RTP receiver wishes to join the
same multicast session, instead of simply issuing a Source Filtering
Group Management Protocol (SFGMP) Join message, it sends a request to
the feedback target for the session and asks for the Reference
Information. The retransmission server starts a new unicast RTP
(retransmission) session and sends the Reference Information to the
RTP receiver over that session. If there is spare bandwidth, the
retransmission server might burst the Reference Information faster
than its natural rate. As soon as the receiver acquires the
Reference Information, it can join the multicast session and start
processing the multicast data. A simplified network diagram showing
this method through an intermediary network element is depicted in
Figure 1.
This method potentially reduces the acquisition delay. We refer to
this method as Unicast-based Rapid Acquisition of Multicast RTP
Sessions. A primary use case for this method is to reduce the
channel-change times in IPTV networks where compressed video streams
are multicast in different SSM sessions and viewers randomly join
these sessions.
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-----------------------
+--->| Intermediary |
| | Network Element |
| ...|(Retransmission Server)|
| : -----------------------
| :
| v
----------- ---------- ----------
| Multicast | | |---------->| Joining |
| Source |------->| Router |..........>| RTP |
| | | | | Receiver |
----------- ---------- ----------
|
| ----------
+---------------->| Existing |
| RTP |
| Receiver |
----------
-------> Multicast RTP Flow
.......> Unicast RTP Flow
Figure 1: Rapid acquisition through an intermediary network element
A principle design goal in this solution is to use the existing tools
in the RTP/RTCP protocol family. This improves the versatility of
the existing implementations, and promotes faster deployment and
better interoperability. To this effect, we use the unicast
retransmission support of RTP [RFC4588] and the capabilities of RTCP
to handle the signaling needed to accomplish the acquisition.
1.1. Acquisition of RTP Streams vs. RTP Sessions
In this memo we describe a protocol that handles the rapid
acquisition of a single multicast RTP session (called primary
multicast RTP session) carrying one or more RTP streams (called
primary multicast streams). If desired, multiple instances of this
protocol may be run in parallel to acquire multiple RTP sessions
simultaneously.
When an RTP receiver requests the Reference Information from the
retransmission server, it can opt to rapidly acquire a specific
subset of the available RTP streams in the primary multicast RTP
session. Alternatively, it can request the rapid acquisition of all
of the RTP streams in that RTP session. Regardless of how many RTP
streams are requested by the RTP receiver or how many will be
actually sent by the retransmission server, only one unicast RTP
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session will be established by the retransmission server. This
unicast RTP session is separate from the associated primary multicast
RTP session. As a result, there are always two different RTP
sessions in a single instance of the rapid acquisition protocol: (i)
the primary multicast RTP session with its associated unicast
feedback and (ii) the unicast RTP session.
If the RTP receiver wants to rapidly acquire multiple RTP sessions
simultaneously, separate unicast RTP sessions will be established for
each of them.
1.2. Outline
In the rest of this specification, we have the following outline: In
Section 4, we describe the delay components in generic multicast
applications. Section 5 presents an overview of the protocol design
considerations for rapid acquisition. We provide the protocol
details of the rapid acquisition method in Section 6 and Section 7.
Section 8 and Section 9 discuss the SDP signaling issues with
examples and NAT-related issues, respectively. Finally, Section 10
discusses the security considerations.
Section 3 provides a list of the definitions frequently used in this
document.
2. Requirements Notation
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 [RFC2119].
3. Definitions
This document uses the following acronyms and definitions frequently:
(Primary) SSM (or Multicast) Session: The multicast session to which
RTP receivers can join at a random point in time. A primary SSM
session can carry multiple RTP streams.
Primary Multicast RTP Session: The multicast RTP session an RTP
receiver is interested in acquiring rapidly. From the RTP receiver's
viewpoint, the primary multicast RTP session has one associated
unicast RTCP feedback stream to a Feedback Target, in addition to the
primary multicast RTP stream(s).
Primary Multicast (RTP) Streams: The RTP stream(s) carried in the
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primary multicast RTP session.
Source Filtering Group Management Protocol (SFGMP): Following the
definition in [RFC4604], SFGMP refers to the Internet Group
Management Protocol (IGMP) version 3 [RFC3376] and the Multicast
Listener Discovery Protocol (MLD) version 2 [RFC3810] in the IPv4 and
IPv6 networks, respectively. However, the rapid acquisition method
introduced in this document does not depend on a specific version of
either of these group management protocols. In the remainder of this
document, SFGMP will refer to any group management protocol that has
Join and Leave functionalities.
Feedback Target (FT): Unicast RTCP feedback target as defined in
[RFC5760]. FT_Ap denotes a specific feedback target running on a
particular address and port.
Retransmission (or Burst) Packet: An RTP packet that is formatted as
defined in Section 4 of [RFC4588]. The payload of a retransmission
or burst packet comprises the retransmission payload header followed
by the payload of the original RTP packet.
Reference Information: The set of certain media content and metadata
information that is sufficient for an RTP receiver to start usefully
consuming a media stream. The meaning, format and size of this
information are specific to the application and are out of scope of
this document.
Preamble Information: A more compact form of the whole or a subset
of the Reference Information transmitted out-of-band.
(Unicast) Burst (or Retransmission) RTP Session: The unicast RTP
session used to send one or more unicast burst RTP streams and their
associated RTCP messages. The terms "burst RTP session" and
"retransmission RTP session" can be used interchangeably.
(Unicast) Burst (Stream): A unicast stream of RTP retransmission
packets that enable an RTP receiver to rapidly acquire the Reference
Information associated with a primary multicast stream. Each burst
stream is identified by its Synchronization Source (SSRC) identifier
that is unique in the primary multicast RTP session. Following the
session-multiplexing guidelines in [RFC4588], each unicast burst
stream will use the same SSRC and CNAME as its primary multicast RTP
stream.
Retransmission Server (RS): The RTP/RTCP endpoint that can generate
the retransmission packets and the burst streams. RS may also
generate other non-retransmission packets to aid rapid acquisition.
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4. Elements of Delay in Multicast Applications
In a source-specific (SSM) multicast delivery system, there are three
major elements that contribute to the overall acquisition delay when
an RTP receiver switches from one multicast session to another one.
These are:
o Multicast switching delay
o Reference Information latency
o Buffering delays
Multicast switching delay is the delay that is experienced to leave
the current multicast session (if any) and join the new multicast
session. In typical systems, the multicast join and leave operations
are handled by a group management protocol. For example, the
receivers and routers participating in a multicast session can use
the Internet Group Management Protocol (IGMP) version 3 [RFC3376] or
the Multicast Listener Discovery Protocol (MLD) version 2 [RFC3810].
In either of these protocols, when a receiver wants to join a
multicast session, it sends a message to its upstream router and the
routing infrastructure sets up the multicast forwarding state to
deliver the packets of the multicast session to the new receiver.
Depending on the proximity of the upstream router, the current state
of the multicast tree, the load on the system and the protocol
implementation, the join times vary. Current systems provide join
latencies usually less than 200 milliseconds (ms). If the receiver
had been participating in another multicast session before joining
the new session, it needs to send a Leave message to its upstream
router to leave the session. In common multicast routing protocols,
the leave times are usually smaller than the join times, however, it
is possible that the Leave and Join messages might get lost, in which
case the multicast switching delay inevitably increases.
Reference Information latency is the time it takes the receiver to
acquire the Reference Information. It is highly dependent on the
proximity of the actual time the receiver joined the session to the
next time the Reference Information will be sent to the receivers in
the session, whether the Reference Information is sent contiguously
or not, and the size of the Reference Information. For some
multicast flows, there is a little or no interdependency in the data,
in which case the Reference Information latency will be nil or
negligible. For other multicast flows, there is a high degree of
interdependency. One example of interest is the multicast flows that
carry compressed audio/video. For these flows, the Reference
Information latency can become quite large and be a major contributor
to the overall delay.
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The buffering component of the overall acquisition delay is driven by
the way the application layer processes the payload. In many
multicast applications, an unreliable transport protocol such as UDP
[RFC0768] is often used to transmit the data packets, and the
reliability, if needed, is usually addressed through other means such
as Forward Error Correction (e.g.,
[I-D.ietf-fecframe-interleaved-fec-scheme]) and retransmission.
These loss-repair methods require buffering at the receiver side to
function properly. In many applications, it is also often necessary
to de-jitter the incoming data packets before feeding them to the
application. The de-jittering process also increases the buffering
delays. Besides these network-related buffering delays, there are
also specific buffering needs that are required by the individual
applications. For example, standard video decoders typically require
an amount, sometimes a significant amount, of coded video data to be
available in the pre-decoding buffers prior to starting to decode the
video bitstream.
5. Protocol Design Considerations and Their Effect on Resource
Management for Rapid Acquisition
This section is for informational purposes and does not contain
requirements for implementations.
Rapid acquisition is an optimization of a system that is expected to
continue to work correctly and properly whether or not the
optimization is effective, or even fails due to lost control and
feedback messages, congestion, or other problems. This is
fundamental to the overall design requirements surrounding the
protocol definition and to the resource management schemes to be
employed together with the protocol (e.g., QoS machinery, server load
management, etc). In particular, the system needs to operate within
a number of constraints:
o First, a rapid acquisition operation must fail gracefully. The
user experience must, except perhaps in pathological
circumstances, be not significantly worse for trying and failing
to complete rapid acquisition compared to simply joining the
multicast session.
o Second, providing the rapid acquisition optimizations must not
cause collateral damage to either the multicast session being
joined, or other multicast sessions sharing resources with the
rapid acquisition operation. In particular, the rapid acquisition
operation must avoid interference with the multicast session that
might be simultaneously being received by other hosts. In
addition, it must also avoid interference with other multicast
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sessions sharing the same network resources. These properties are
possible, but are usually difficult to achieve.
One challenge is the existence of multiple bandwidth bottlenecks
between the receiver and the server(s) in the network providing the
rapid acquisition service. In commercial IPTV deployments, for
example, bottlenecks are often present in the aggregation network
connecting the IPTV servers to the network edge, the access links
(e.g., DSL, DOCSIS) and in the home network of the subscribers. Some
of these links might serve only a single subscriber, limiting
congestion impact to the traffic of only that subscriber, but others
can be shared links carrying multicast sessions of many subscribers.
Also note that the state of these links can vary over time. The
receiver might have knowledge of a portion of this network, or might
have partial knowledge of the entire network. The methods employed
by the devices to acquire this network state information is out of
scope for this document. The receiver should be able to signal the
server with the bandwidth that it believes it can handle. The server
also needs to be able to rate limit the flow in order to stay within
the performance envelope that it knows about. Both the server and
receiver need to be able to inform the other of changes in the
requested and delivered rates. However, the protocol must be robust
in the presence of packet loss, so this signaling must include the
appropriate default behaviors.
A second challenge is that for some uses (e.g., high-bitrate video)
the unicast burst bitrate is high while the flow duration of the
unicast burst is short. This is because the purpose of the unicast
burst is to allow the RTP receiver to join the multicast quickly and
thereby limit the overall resources consumed by the burst. Such
high-bitrate, short-duration flows are not amenable to conventional
admission control techniques. For example, end-to-end per-flow
signaled admission control techniques such as RSVP have too much
latency and control channel overhead to be a good fit for rapid
acquisition. Similarly, using a TCP (or TCP-like) approach with a
3-way handshake and slow-start to avoid inducing congestion would
defeat the purpose of attempting rapid acquisition in the first place
by introducing many round-trip times (RTT) of delay.
These observations lead to certain unavoidable requirements and goals
for a rapid acquisition protocol. These are:
o The protocol must be designed to allow a deterministic upper bound
on the extra bandwidth used (compared to just joining the
multicast session). A reasonable size bound is e*B, where B is
the nominal bandwidth of the primary multicast streams, and e is
an excess-bandwidth coefficient. The total duration of the
unicast burst must have a reasonable bound; long unicast bursts
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devolve to the bandwidth profile of multi-unicast for the whole
system.
o The scheme should minimize (or better eliminate) the overlap of
the unicast burst and the primary multicast stream. This
minimizes the window during which congestion could be induced on a
bottleneck link compared to just carrying the multicast or unicast
packets alone.
o The scheme must minimize (or better eliminate) any gap between the
unicast burst and the primary multicast stream, which has to be
repaired later, or in the absence of repair, will result in loss
being experienced by the application.
In addition to the above, there are some other protocol design issues
to be considered. First, there is at least one RTT of "slop" in the
control loop. In starting a rapid acquisition burst, this manifests
as the time between the client requesting the unicast burst and the
burst description and/or the first unicast burst packets arriving at
the receiver. For managing and terminating the unicast burst, there
are two possible approaches for the control loop: The receiver can
adapt to the unicast burst as received, converge based on observation
and explicitly terminate the unicast burst with a second control loop
exchange (which takes a minimum of one RTT, just as starting the
unicast burst does). Alternatively, the server generating the
unicast burst can pre-compute the burst parameters based on the
information in the initial request and tell the receiver the burst
duration.
The protocol described in the next section allows either method of
controlling the rapid acquisition unicast burst.
6. Rapid Acquisition of Multicast RTP Sessions (RAMS)
We start this section with an overview of the rapid acquisition of
multicast sessions (RAMS) method.
6.1. Overview
[RFC5760] specifies an extension to the RTP Control Protocol (RTCP)
to use unicast feedback in an SSM session. It defines an
architecture that introduces the concept of Distribution Source,
which - in an SSM context - distributes the RTP data and
redistributes RTCP information to all RTP receivers. This RTCP
information is retrieved from the Feedback Target, to which RTCP
unicast feedback traffic is sent. One or more entities different
from the Distribution Source MAY implement the feedback target
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functionality, and different RTP receivers MAY use different feedback
targets.
This document builds further on these concepts to reduce the
acquisition delay when an RTP receiver joins a multicast session at a
random point in time by introducing the concept of the Burst Source
and new RTCP feedback messages. The Burst Source has a cache where
the most recent packets from the primary multicast RTP session are
continuously stored. When an RTP receiver wants to receive a primary
multicast stream, it can first request a unicast burst from the Burst
Source before it joins the SSM session. In this burst, the packets
are formatted as RTP retransmission packets [RFC4588] and carry
Reference Information. This information allows the RTP receiver to
start usefully consuming the RTP packets sent in the primary
multicast RTP session.
Using an accelerated bitrate (as compared to the nominal bitrate of
the primary multicast stream) for the unicast burst implies that at a
certain point in time, the payload transmitted in the unicast burst
is going to be the same as the payload in the associated multicast
stream, i.e., the unicast burst will catch up with the primary
multicast stream. At this point, the RTP receiver no longer needs to
receive the unicast burst and can join the SSM session. This method
is referred to as the Rapid Acquisition of Multicast Sessions (RAMS).
This document proposes extensions to [RFC4585] for an RTP receiver to
request a unicast burst as well as for additional control messaging
that can be leveraged during the acquisition process.
6.2. Message Flows
Figure 2 shows the main entities involved in rapid acquisition and
the message flows. They are
o Multicast Source
o Feedback Target (FT)
o Burst/Retransmission Source (BRS)
o RTP Receiver (RTP_Rx)
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----------- --------------
| |------------------------------------>| |
| |.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.->| |
| | | |
| Multicast | ---------------- | |
| Source | | Retransmission | | |
| |-------->| Server (RS) | | |
| |.-.-.-.->| | | |
| | | ------------ | | |
----------- | | Feedback | |<.=.=.=.=.| |
| | Target (FT)| |<~~~~~~~~~| RTP Receiver |
PRIMARY MULTICAST | ------------ | | (RTP_Rx) |
RTP SESSION with | | | |
UNICAST FEEDBACK | | | |
| | | |
- - - - - - - - - - - |- - - - - - - - |- - - - - |- - - - - - - |- -
| | | |
UNICAST BURST | ------------ | | |
(or RETRANSMISSION) | | Burst and | |<~~~~~~~~>| |
RTP SESSION | | Retrans. | |.........>| |
| |Source (BRS)| |<.=.=.=.=>| |
| ------------ | | |
| | | |
---------------- --------------
-------> Multicast RTP Flow
.-.-.-.> Multicast RTCP Flow
.=.=.=.> Unicast RTCP Reports
~~~~~~~> Unicast RTCP Feedback Messages
.......> Unicast RTP Flow
Figure 2: Flow diagram for unicast-based rapid acquisition
The feedback target (FT) is the entity as defined in [RFC5760], to
which RTP_Rx sends its RTCP feedback messages indicating packet loss
by means of an RTCP NACK message or indicating RTP_Rx's desire to
rapidly acquire the primary multicast RTP session by means of an RTCP
feedback message defined in this document. While the Burst/
Retransmission Source (BRS) is responsible for responding to these
messages and for further RTCP interaction with RTP_Rx in the case of
a rapid acquisition process, it is assumed in the remainder of the
document that these two logical entities (FT and BRS) are combined in
a single physical entity and they share state. In the remainder of
the text, the term Retransmission Server (RS) is used whenever
appropriate, to refer to this single physical entity.
FT is involved in the primary multicast RTP session and receives
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unicast feedback for that session as in [RFC5760]. BRS is involved
in the unicast burst (or retransmission) RTP session and transmits
the unicast burst and retransmission packets formatted as RTP
retransmission packets [RFC4588] in a single separate unicast RTP
session to each RTP_Rx. In the unicast burst RTP session, as in any
other RTP session, the BRS and RTP_Rx regularly send the periodic
sender and receiver reports, respectively.
The unicast burst is started by an RTCP feedback message that is
defined in this document based on the common packet format provided
in [RFC4585]. An RTP retransmission is triggered by an RTCP NACK
message defined in [RFC4585]. Both of these messages are sent to FT
of the primary multicast RTP session, and can start the unicast
burst/retransmission RTP session.
In the RTP/AVPF profile, there are certain rules that apply to
scheduling of both of these messages sent to FT in the primary
multicast RTP session, and these are detailed in Section 3.5 of
[RFC4585]. One of the rules states that in a multi-party session
such as the SSM sessions we are considering in this specification, an
RTP_Rx cannot send an RTCP feedback message for a minimum of one
second period after joining the session (i.e., Tmin=1.0 second).
While this rule has the goal of avoiding problems associated with
flash crowds in typical multi-party sessions, it defeats the purpose
of rapid acquisition. Furthermore, when RTP receivers delay their
messages requesting a burst by a deterministic or even a random
amount, it still does not make a difference since such messages are
not related and their timings are independent from each other. Thus,
in this specification we initialize Tmin to zero and allow the RTP
receivers to send a burst request message right at the beginning.
For the subsequent messages during rapid acquisition, the timing
rules of [RFC4585] still apply.
Figure 3 depicts an example of messaging flow for rapid acquisition.
The RTCP feedback messages are explained below. The optional
messages are indicated in parentheses and they might or might not be
present during rapid acquisition. In a scenario where rapid
acquisition is performed by a feedback target co-located with the
media sender, the same method (with the identical message flows)
still applies.
-------------------------
| Retransmission Server |
----------- | ------ ------------ | -------- ------------
| Multicast | | | FT | | Burst/Ret. | | | | | RTP |
| Source | | | | | Source | | | Router | | Receiver |
| | | ------ ------------ | | | | (RTP_Rx) |
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----------- | | | | -------- ------------
| ------------------------- | |
| | | | |
|-- RTP Multicast ---------->--------------->| |
|-. RTCP Multicast -.-.-.-.->-.-.-.-.-.-.-.->| |
| | | | |
| | |********************************|
| | |* PORT MAPPING SETUP *|
| | |********************************|
| | | | |
| |<~~~~~~~~~~~~~~~~~~~~~~~~~ RTCP RAMS-R ~~~|
| | | | |
| | |********************************|
| | |* UNICAST SESSION ESTABLISHED *|
| | |********************************|
| | | | |
| | |~~~ RTCP RAMS-I ~~~~~~~~~~~~~~~>|
| | | | |
| | |... Unicast RTP Burst .........>|
| | | | |
| |<~~~~~~~~~~~~~~~~~~~~~~~~ (RTCP RAMS-R) ~~|
| | | | |
| | |~~ (RTCP RAMS-I) ~~~~~~~~~~~~~~>|
| | | | |
| | | | |
| | | |<= SFGMP Join ==|
| | | | |
|-- RTP Multicast ------------------------------------------->|
|-. RTCP Multicast -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.>|
| | | | |
| | | | |
| | |<~~~~~~~~~~~~~~~ RTCP RAMS-T ~~~|
| | | | |
| | | | |
| | | | |
: : : : :
: : : : :
| | |<.=.= Unicast RTCP Reports .=.=>|
: : : (for the unicast session) :
: : : : :
| | | | |
-------> Multicast RTP Flow
.-.-.-.> Multicast RTCP Flow
.=.=.=.> Unicast RTCP Reports
~~~~~~~> Unicast RTCP Feedback Messages
=======> SFGMP Messages
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.......> Unicast RTP Flow
Figure 3: Message flows for unicast-based rapid acquisition
This document defines the expected behaviors of RS and RTP_Rx. It is
instructive to have independently operating implementations on RS and
RTP_Rx that request the burst, describe the burst, start the burst,
join the multicast session and stop the burst. These implementations
send messages to each other, but provisions are needed for the cases
where the control messages get lost, or re-ordered, or are not being
delivered to their destinations.
The following steps describe rapid acquisition in detail:
1. Port Mapping Setup: For the primary multicast RTP session, the
RTP and RTCP destination ports are declaratively specified
(Refer to Section 8 for examples in SDP). However, RTP_Rx needs
to choose its RTP and RTCP receive ports in the unicast burst
RTP session. Since this unicast session is established after
RTP_Rx has sent a RAMS-Request (RAMS-R) message as unicast
feedback in the primary multicast RTP session, RTP_Rx MUST first
setup the port mappings between the unicast and multicast
sessions and send this mapping information to FT along with the
RAMS-R message so that BRS knows how to communicate with RTP_Rx.
The details of this setup procedure are explained in
[I-D.ietf-avt-ports-for-ucast-mcast-rtp]. Other NAT-related
issues are left to Section 9 to keep the present discussion
focused on the RAMS message flows.
2. Request: RTP_Rx sends a rapid acquisition request (RAMS-R) for
the primary multicast RTP session to the unicast feedback target
of that session. The request contains the SSRC identifier of
RTP_Rx (aka SSRC of packet sender) and can contain the media
sender SSRC identifier(s) of the primary multicast stream(s) of
interest (aka SSRC of media source). The RAMS-R message can
contain parameters that constrain the burst, such as the buffer
and bandwidth limits.
Before joining the SSM session, RTP_Rx learns the addresses for
the multicast source, group and RS by out-of-band means. If
RTP_Rx desires to rapidly acquire only a subset of the primary
multicast streams available in the primary multicast RTP
session, RTP_Rx MUST also acquire the SSRC identifiers for the
desired RTP streams out-of-band. Based on this information,
RTP_Rx populates the desired SSRC(s) in the RAMS-R message.
When FT successfully receives the RAMS-R message, BRS responds
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to it by accepting or rejecting the request. Immediately before
BRS sends any RTP or RTCP packet(s) described below, it
establishes the unicast burst RTP session.
3. Response: BRS sends RAMS-Information (RAMS-I) message(s) to
RTP_Rx to convey the status for the burst(s) requested by
RTP_Rx.
If the primary multicast RTP session associated with FT_Ap on
which the RAMS-R message was received contains only a single
primary multicast stream, BRS SHALL always use the SSRC of the
RTP stream associated with FT_Ap in the RAMS-I message(s)
regardless of the media sender SSRC requested in the RAMS-R
message. In such cases the 'ssrc' attribute MAY be omitted from
the media description. If the requested SSRC and the actual
media sender SSRC do not match, BRS MUST explicitly populate the
correct media sender SSRC in the initial RAMS-I message (See
Section 7.3).
FT_Ap could also be associated with an RTP session that carries
two or more primary multicast streams. If RTP_Rx will issue a
collective request to receive the whole primary multicast RTP
session, it does not need the 'ssrc' attributes to be described
in the media description.
If FT_Ap is associated with two or more RTP sessions, RTP_Rx's
request will be ambiguous. To avoid any ambiguity, each FT_Ap
MUST only associate itself with a single RTP session.
If RTP_Rx is willing to rapidly acquire only a subset of the
primary multicast streams, RTP_Rx MUST list all the media sender
SSRC(s) denoting the stream(s) it wishes to acquire in the
RAMS-R message. Upon receiving such a message, BRS MAY accept
the request for all or a subset of the media sender SSRC(s) that
matched the RTP stream(s) it serves. BRS MUST reject all other
requests with an appropriate response code.
* Reject Responses: BRS MUST take into account any limitations
that may have been specified by RTP_Rx in the RAMS-R message
when making a decision regarding the request. If RTP_Rx has
requested to acquire the whole primary multicast RTP session
but BRS cannot provide a rapid acquisition service for any of
the primary multicast streams, BRS MUST reject the request
via a single RAMS-I message with a collective reject response
code and whose media sender SSRC field is set to one of SSRCs
served by this FT_Ap. Upon receiving this RAMS-I message,
RTP_Rx abandons the rapid acquisition attempt and can
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immediately join the multicast session by sending an SFGMP
Join message towards its upstream multicast router.
In all other cases, BRS MUST send a separate RAMS-I message
with the appropriate response code for each primary multicast
stream that has been requested by RTP_Rx but cannot be served
by BRS.
* Accept Responses: BRS MUST send at least one separate RAMS-I
message with the appropriate response code for each primary
multicast stream that has been requested by RTP_Rx and will
be served by BRS. Such RAMS-I messages comprise fields that
can be used to describe the individual unicast burst streams.
When there is a RAMS-R request for multiple primary multicast
streams, BRS MUST include all the individual RAMS-I messages
corresponding to that RAMS-R request in the same compound
RTCP packet if these messages fit in the same packet.
The RAMS-I message carries the RTP sequence number of the
first packet transmitted in the respective RTP stream to
allow RTP_Rx to detect any missing initial packet(s). When
BRS accepts a request for a primary multicast stream, this
field MUST always be populated in the RAMS-I message(s) sent
for this particular primary multicast stream. It is
RECOMMENDED that BRS sends a RAMS-I message at the start of
the burst so that RTP_Rx can quickly detect any missing
initial packet(s).
It is possible that the RAMS-I message for a primary multicast
stream can get delayed or lost, and RTP_Rx can start receiving
RTP packets before receiving a RAMS-I message. RTP_Rx MUST NOT
make protocol dependencies on quickly receiving the initial
RAMS-I message. For redundancy purposes, it is RECOMMENDED that
BRS repeats the RAMS-I messages multiple times as long as it
follows the RTCP timer rules defined in [RFC4585].
4. Unicast Burst: For the primary multicast stream(s) for which
the request is accepted, BRS starts sending the unicast burst(s)
that comprises one or more RTP retransmission packets sent in
the unicast burst RTP session. In addition, BRS MAY send
preamble information data to RTP_Rx in addition to the requested
burst, to prime the media decoder(s). The format of this
preamble data is RTP-payload specific and not specified here.
5. Updated Request: RTP_Rx MAY send an updated RAMS-R message (as
unicast feedback in the primary multicast RTP session) with a
different value for one or more fields of an earlier RAMS-R
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message. If there is already a burst planned for or being
transmitted to a particular RTP_Rx for a particular stream, the
newly arriving RAMS-R is an updated request; otherwise, it is a
new request. BRS determines the identity of the requesting
RTP_Rx by looking at its canonical name identifier (CNAME item
in the SDES source description). Thus, RTP_Rx MUST choose a
globally unique CNAME identifier. Different such ways are
provided in [I-D.ietf-avt-rtp-cnames]. In addition to one or
more fields with updated values, an updated RAMS-R message may
also include the fields whose values did not change.
Upon receiving an updated request, BRS can use the updated
values for sending/shaping the burst, or refine the values and
use the refined values for sending/shaping the burst.
Subsequently, BRS MAY send an updated RAMS-I message in the
unicast burst RTP session to indicate the changes it made.
RTP_Rx may be in an environment where the available resources
are time-varying, which may or may not deserve sending a new
updated request. Determining the circumstances where RTP_Rx
needs or does not need to send an updated request and the
methods that RTP_Rx can use to detect and evaluate the time-
varying available resources are not specified in this document.
6. Updated Response: BRS can send more than one RAMS-I messages in
the unicast burst RTP session, e.g., to update the value of one
or more fields in an earlier RAMS-I message. The updated RAMS-I
messages might or might not be a direct response to a RAMS-R
message. BRS can also send updated RAMS-I messages to signal
RTP_Rx in real time to join the SSM session, when BRS had
already sent an initial RAMS-I message, e.g., at the start of
the burst. RTP_Rx depends on BRS to learn the join time, which
can be conveyed by the first RAMS-I message, or can be sent/
revised in a later RAMS-I message. If BRS is not capable of
determining the join time in the initial RAMS-I message, BRS
MUST send another RAMS-I message (with the join time
information) later.
7. Multicast Join Signaling: The RAMS-I message allows BRS to
signal explicitly when RTP_Rx needs to send the SFGMP Join
message. RTP_Rx SHOULD use this information from the most
recent RAMS-I message unless it has more accurate information.
If the request is accepted, this information MUST be conveyed in
at least one RAMS-I message and its value MAY be updated by
subsequent RAMS-I messages.
There can be missing packets if RTP_Rx joins the multicast
session too early or too late. For example, if RTP_Rx starts
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receiving the primary multicast stream while it is still
receiving the unicast burst at a high excess bitrate, this can
result in an increased risk of packet loss. Or, if RTP_Rx joins
the multicast session some time after the unicast burst is
finished, there can be a gap between the burst and multicast
data (a number of RTP packets might be missing). In both cases,
RTP_Rx can issue retransmission requests (via RTCP NACK messages
sent as unicast feedback in the primary multicast RTP session)
[RFC4585] to the FT entity of RS to fill the gap. BRS might or
might not respond to such requests. When it responds and the
response causes significant changes in one or more values
reported earlier to RTP_Rx, an updated RAMS-I SHOULD be sent to
RTP_Rx.
8. Multicast Receive: After the join, RTP_Rx starts receiving the
primary multicast stream(s).
9. Terminate: BRS can know when it needs to ultimately stop the
unicast burst based on its parameters. However, RTP_Rx may need
to ask BRS to terminate the burst prematurely or at a specific
sequence number. For this purpose, it uses the RAMS-Termination
(RAMS-T) message sent as RTCP feedback in the unicast burst RTP
session. A separate RAMS-T message is sent for each primary
multicast stream served by BRS unless an RTCP BYE message has
been sent in the unicast burst RTP session as described in Step
10. For the burst requests that were rejected by BRS, there is
no need to send a RAMS-T message.
If RTP_Rx wants to terminate a burst prematurely, it SHALL send
a plain RAMS-T message for the SSRC of the primary multicast
stream it wishes to terminate. This message is sent in the
unicast burst RTP session. Upon receiving this message BRS MUST
terminate the unicast burst. If RTP_Rx requested to acquire the
entire primary multicast RTP session but wants to terminate this
request before it learns the individual media sender SSRC(s) via
RAMS-I message(s) or RTP packets, RTP_Rx cannot use RAMS-T
message(s) and thus MUST send an RTCP BYE message in the unicast
burst RTP session to terminate the request.
Otherwise, the default behavior for RTP_Rx is to send a RAMS-T
message in the unicast burst RTP session immediately after it
joins the multicast session and started receiving multicast
packets. In that case, RTP_Rx SHALL send a RAMS-T message with
the sequence number of the first RTP packet received in the
primary multicast stream. Then, BRS MUST terminate the
respective burst after it sends the unicast burst packet whose
Original Sequence Number (OSN) field in the RTP retransmission
payload header matches this number minus one.
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If an RTCP BYE message has not been issued yet as described in
Step 10, RTP_Rx MUST send at least one RAMS-T message for each
primary multicast stream served by BRS. The RAMS-T message(s)
MUST be addressed to BRS and sent in the unicast burst RTP
session. Against the possibility of a message loss, it is
RECOMMENDED that RTP_Rx repeats the RAMS-T messages multiple
times as long as it follows the RTCP timer rules defined in
[RFC4585].
The binding between RAMS-T and ongoing bursts is achieved
through RTP_Rx's CNAME identifier, and packet sender and media
sender SSRCs. Choosing a globally unique CNAME makes sure that
the RAMS-T messages are processed correctly.
10. Terminate with RTCP BYE: When RTP_Rx is receiving one or more
burst streams, if RTP_Rx becomes no longer interested in
acquiring any of the primary multicast streams, RTP_Rx SHALL
issue an RTCP BYE message for the unicast burst RTP session and
another RTCP BYE message for the primary multicast RTP session.
These RTCP BYE messages are sent to BRS and FT logical entities,
respectively.
Upon receiving an RTCP BYE message, BRS MUST terminate the rapid
acquisition operation, and cease transmitting any further burst
packets and retransmission packets. If support for [RFC5506]
has been signaled, the RTCP BYE message MAY be sent in a
reduced-size RTCP packet. Otherwise, Section 6.1 of [RFC3550]
mandates the RTCP BYE message always to be sent with a sender or
receiver report in a compound RTCP packet. If no data has been
received, an empty receiver report MUST be still included. With
the information contained in the receiver report, RS can figure
out how many duplicate RTP packets have been delivered to RTP_Rx
(Note that this will be an upper-bound estimate as one or more
packets might have been lost during the burst transmission).
The impact of duplicate packets and measures that can be taken
to minimize the impact of receiving duplicate packets will be
addressed in Section 6.4.
Since an RTCP BYE message issued for the unicast burst RTP
session terminates that session and ceases transmitting any
further packets in that session, there is no need for sending
explicit RAMS-T messages, which would only terminate their
respective bursts.
For the purpose of gathering detailed information about RTP_Rx's
rapid acquisition experience, [I-D.ietf-avt-multicast-acq-rtcp-xr]
defines an RTCP Extended Report (XR) Block. This report is designed
to be payload-independent, thus, it can be used by any multicast
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application that supports rapid acquisition. Support for this XR
report is, however, OPTIONAL.
6.3. Synchronization of Primary Multicast Streams
When RTP_Rx acquires multiple primary multicast streams, RTP_Rx can
need to synchronize them for the playout. This synchronization is
traditionally achieved by the help of the RTCP sender reports
[RFC3550]. If the playout will start before RTP_Rx has joined the
multicast session, RTP_Rx needs to receive the information reflecting
the synchronization among the primary multicast streams early enough
so that it can play out the media in a synchronized fashion.
The suggested approach is to use the RTP header extension mechanism
[RFC5285] and convey the synchronization information in a header
extension as defined in [I-D.ietf-avt-rapid-rtp-sync]. Per [RFC4588]
"if the original RTP header carried an RTP header extension, the
retransmission packet SHOULD carry the same header extension." Thus,
as long as the multicast source emits a header extension with the
synchronization information frequently enough, there is no additional
task that needs to be carried out by BRS. The synchronization
information will be sent to RTP_Rx along with the burst packets. The
frequent header extensions sent in the primary multicast RTP sessions
also allow rapid synchronization of the RTP streams for the RTP
receivers that do not support RAMS or that directly join the
multicast session without running RAMS. Thus, in RAMS applications,
it is RECOMMENDED that the multicast sources frequently send
synchronization information by using header extensions following the
rules presented in [I-D.ietf-avt-rapid-rtp-sync]. The regular sender
reports are still sent in the unicast session by following the rules
of [RFC3550].
6.4. Burst Shaping and Congestion Control in RAMS
This section provides informative guidelines about how BRS can shape
the transmission of the unicast burst and how congestion can be dealt
within the RAMS process. When RTP_Rx detects that the unicast burst
is causing severe congestion, it can prefer to send a RAMS-T message
immediately to stop the unicast burst.
A higher bitrate for the unicast burst naturally conveys the
Reference Information and media content to RTP_Rx faster. This way,
RTP_Rx can start consuming the data sooner, which results in a faster
acquisition. A higher bitrate also represents a better utilization
of BRS resources. As the burst may continue until it catches up with
the primary multicast stream, the higher the bursting bitrate, the
less data BRS needs to transmit. However, a higher bitrate for the
burst also increases the chances for congestion-caused packet loss.
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Thus, as discussed in Section 5, there has to be an upper bound on
the bandwidth used by the burst.
When BRS transmits the unicast burst, it is expected to take into
account all available information to prevent any packet loss that
might take place during the bursting as a result of buffer overflow
on the path between RS and RTP_Rx and at RTP_Rx itself. The bursting
bitrate can be determined by taking into account the following
information, when available:
a. Information obtained via the RAMS-R message, such as Max RAMS
Buffer Fill Requirement and/or Max Receive Bitrate (See
Section 7.2).
b. Information obtained via RTCP receiver reports provided by RTP_Rx
in the retransmission session, allowing in-session bitrate
adaptations for the burst. When these receiver reports indicate
packet loss, this can indicate a certain congestion state in the
path from RS to RTP_Rx.
c. Information obtained via RTCP NACKs provided by RTP_Rx in the
primary multicast RTP session, allowing in-session bitrate
adaptations for the burst. Such RTCP NACKs are transmitted by
RTP_Rx in response to packet loss detection in the burst. NACKs
can indicate a certain congestion state on the path from RS to
RTP_Rx.
d. There can be other feedback received from RTP_Rx, e.g., in the
form of ECN-CE markings [I-D.ietf-avt-ecn-for-rtp] that can
influence in-session bitrate adaptation.
e. Information obtained via updated RAMS-R messages, allowing in-
session bitrate adaptations, if supported by BRS.
f. Transport protocol-specific information. For example, when DCCP
is used to transport the RTP burst, the ACKs from the DCCP client
can be leveraged by the BRS / DCCP server for burst shaping and
congestion control.
g. Pre-configured settings for each RTP_Rx or a set of RTP_Rxs that
indicate the upper-bound bursting bitrates for which no packet
loss will occur as a result of congestion along the path of RS to
RTP_Rx. For example, in managed IPTV networks, where the
bottleneck bandwidth along the end-to-end path is known and where
the network between RS and this link is provisioned and
dimensioned to carry the burst streams, the bursting bitrate does
not exceed the provisioned value. These settings can also be
dynamically adapted using application-aware knowledge.
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BRS chooses the initial burst bitrate as follows:
o When using RAMS in environments as described in (g), BRS MUST
transmit the burst packets at an initial bitrate higher than the
nominal bitrate, but within the engineered or reserved bandwidth
limit.
o When BRS cannot determine a reliable bitrate value for the unicast
burst (using a through g), it is desirable that BRS chooses an
appropriate initial bitrate not above the nominal bitrate and
increases it gradually unless a congestion is detected.
In both cases, during the burst transmission BRS MUST continuously
monitor for packet losses as a result of congestion by means of one
or more of the mechanisms described in (b,c,d,e,f). When BRS relies
on RTCP receiver reports, sufficient bandwidth needs to be provided
to RTP Rx for RTCP transmission in the unicast burst RTP session. To
achieve a reasonable fast adaptation against congestion, it is
recommended that RTP_Rx sends a receiver report at least once every
two RTTs between RS and RTP_Rx. Although the specific heuristics and
algorithms that deduce a congestion state and how subsequently BRS
acts are outside the scope of this specification, the following two
methods are the best practices:
o Upon detection of a significant amount of packet loss, which BRS
attributes to congestion, BRS decreases the burst bitrate. The
rate by which BRS increases and decreases the bitrate for the
burst can be determined by a TCP-friendly bitrate adaptation
algorithm for RTP over UDP , or in the case of (f) by the
congestion control algorithms defined in DCCP [RFC5762].
o If the congestion is persistent and BRS has to reduce the burst
bitrate to a point where the RTP Rx buffer might underrun or the
burst will consume too many resources, BRS terminates the burst
and transmits a RAMS-I message to RTP Rx with the appropriate
response code. It is then up to RTP Rx to decide when to join the
multicast session.
Even though there is no congestion experienced during the burst,
congestion may anyway arise near the end of the burst as RTP_Rx
eventually needs to join the multicast session. During this brief
period both the burst packets and the multicast packets can be
simultaneously received by RTP_Rx, thus enhancing the risk of
congestion.
Since BRS signals RTP_Rx when BRS expects RTP_Rx to send the SFGMP
Join message, BRS can have a rough estimate of when RTP_Rx will start
receiving multicast packets in the SSM session. BRS can keep on
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sending burst packets but reduces the bitrate accordingly at the
appropriate instant by taking the bitrate of the whole SSM session
into account. If BRS ceases transmitting the burst packets before
the burst catches up, any gap resulting from this imperfect switch-
over by RTP_Rx can be later repaired by requesting retransmissions
for the missing packets from RS. The retransmissions can be shaped
by BRS to make sure that they do not cause collateral loss in the
primary multicast RTP session and the unicast burst RTP session.
6.5. Failure Cases
In the following, we examine the implications of losing the RAMS-R,
RAMS-I or RAMS-T messages and other failure cases.
When RTP_Rx sends a RAMS-R message to initiate a rapid acquisition
but the message gets lost and FT does not receive it, RTP_Rx will get
neither a RAMS-I message, nor a unicast burst. In this case, RTP_Rx
MAY resend the request when it is eligible to do so based on the RTCP
timer rules defined in [RFC4585]. Or, after a reasonable amount of
time, RTP_Rx can time out (based on the previous observed response
times) and immediately join the SSM session.
In the case RTP_Rx starts receiving a unicast burst but it does not
receive a corresponding RAMS-I message within a reasonable amount of
time, RTP_Rx can either discard the burst data or decide not to
interrupt the unicast burst, and be prepared to join the SSM session
at an appropriate time it determines or as indicated in a subsequent
RAMS-I message (if available). If the network is subject to packet
loss, it is RECOMMENDED that BRS repeats the RAMS-I messages multiple
times based on the RTCP timer rules defined in [RFC4585].
In the failure cases where the RAMS-R message is lost and RTP_Rx
gives up, or the RAMS-I message is lost, RTP_Rx MUST still terminate
the burst(s) it requested by following the rules described in
Section 6.2.
In the case a RAMS-T message sent by RTP_Rx does not reach its
destination, BRS can continue sending burst packets even though
RTP_Rx no longer needs them. In such cases, it is RECOMMENDED that
RTP_Rx repeats the RAMS-T message multiple times based on the RTCP
timer rules defined in [RFC4585]. BRS MUST be provisioned to
deterministically terminate the burst when it can no longer send the
burst packets faster than it receives the primary multicast stream
packets.
Section 6.3.5 of [RFC3550] explains the rules pertaining to timing
out an SSRC. When BRS accepts to serve the requested burst(s) and
establishes the retransmission session, BRS needs to check the
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liveness of RTP_Rx via the RTCP messages and reports RTP_Rx sends.
The default rules explained in [RFC3550] apply in RAMS as well.
7. Encoding of the Signaling Protocol in RTCP
This section defines the formats of the RTCP transport-layer feedback
messages that are exchanged between the retransmission server (RS)
and RTP receiver (RTP_Rx) during rapid acquisition. These messages
are referred to as the RAMS Messages. They are payload-independent
and MUST be used by all RTP-based multicast applications that support
rapid acquisition regardless of the payload they carry.
Payload-specific feedback messages are not defined in this document.
However, further optional payload-independent and payload-specific
information can be included in the exchange.
The common packet format for the RTCP feedback messages is defined in
Section 6.1 of [RFC4585] and is also sketched in Figure 4.
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| FMT | PT | length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of packet sender |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of media source |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: Feedback Control Information (FCI) :
: :
Figure 4: The common packet format for the RTCP feedback messages
Each feedback message has a fixed-length field for version, padding,
feedback message type (FMT), payload type (PT), length, SSRC of
packet sender, SSRC of media sender as well as a variable-length
field for feedback control information (FCI).
In the RAMS messages, the PT field is set to RTPFB (205) and the FMT
field is set to RAMS (6). Individual RAMS messages are identified by
a sub-field called Sub Feedback Message Type (SFMT). Any Reserved
field SHALL be set to zero and ignored.
Depending on the specific scenario and timeliness/importance of a
RAMS message, it can be desirable to send it in a reduced-size RTCP
packet [RFC5506]. However, unless support for [RFC5506] has been
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signaled, compound RTCP packets MUST be used by following [RFC3550]
rules.
Following the rules specified in [RFC3550], all integer fields in the
messages defined below are carried in network-byte order, that is,
most significant byte (octet) first, also known as big-endian.
Unless otherwise stated, numeric constants are in decimal (base 10).
7.1. Extensions
To improve the functionality of the RAMS method in certain
applications, it can be desirable to define new fields in the RAMS
Request, Information and Termination messages. Such fields MUST be
encoded as Type-Length-Value (TLV) elements as described below and
sketched in Figure 5:
o Type: A single-octet identifier that defines the type of the
parameter represented in this TLV element.
o Length: A two-octet field that indicates the length (in octets)
of the TLV element excluding the Type and Length fields, and the
8-bit Reserved field between them. This length does not include
any padding that is required for alignment.
o Value: Variable-size set of octets that contains the specific
value for the parameter.
In the extensions, the Reserved field SHALL be set to zero and
ignored. If a TLV element does not fall on a 32-bit boundary, the
last word MUST be padded to the boundary using further bits set to
zero.
In a RAMS message, any vendor-neutral or private extension MUST be
placed after the mandatory fields and mandatory TLV elements (if
any). The extensions MAY be placed in any order. The support for
extensions (unless specified otherwise) is OPTIONAL.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Type | Reserved | Length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: Value :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 5: Structure of a TLV element
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7.1.1. Vendor-Neutral Extensions
If the goal in defining new TLV elements is to extend the
functionality in a vendor-neutral manner, they MUST be registered
with IANA through the guidelines provided in Section 11.5.
The current document defines several vendor-neutral extensions in the
subsequent sections.
7.1.2. Private Extensions
It is desirable to allow vendors to use private extensions in a TLV
format. For interoperability, such extensions must not collide with
each other.
A certain range of TLV Types (between - and including - 128 and 254 )
is reserved for private extensions (Refer to Section 11.5). IANA
management for these extensions is unnecessary and they are the
responsibility of individual vendors.
The structure that MUST be used for the private extensions is
depicted in Figure 6. Here, the enterprise numbers are used from
http://www.iana.org/assignments/enterprise-numbers. This will ensure
the uniqueness of the private extensions and avoid any collision.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Type | Reserved | Length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Enterprise Number |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: Value :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 6: Structure of a private extension
7.2. RAMS Request
The RAMS Request message is identified by SFMT=1. This message is
sent as unicast feedback in the primary multicast RTP session by
RTP_Rx to request rapid acquisition of a primary multicast RTP
session, or one or more primary multicast streams belonging to the
same primary multicast RTP session. In the RAMS-R message, RTP_Rx
MUST set both the packet sender SSRC and media sender SSRC fields to
its own SSRC since the media sender SSRC may not be known. RTP_Rx
MUST provide explicit signaling as described below to request
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stream(s). This minimizes the chances of accidentally requesting a
wrong primary multicast stream.
The FCI field MUST contain only one RAMS Request. The FCI field has
the structure depicted in Figure 7.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SFMT=1 | Reserved |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: Optional TLV-encoded Fields (and Padding, if needed) :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 7: FCI field syntax for the RAMS Request message
o Requested Media Sender SSRC(s): Mandatory TLV element that lists
the media sender SSRC(s) requested by RTP_Rx. BRS MUST ignore the
media sender SSRC specified in the header of the RAMS-R message.
If the Length field is set to zero (i.e., no media sender SSRC is
listed), it means that RTP_Rx is requesting to rapidly acquire the
entire primary multicast RTP session. Otherwise, RTP_Rx lists the
individual media sender SSRCs in this TLV element and sets the
Length field of the TLV element to 4*n, where n is the number of
SSRC entries.
Type: 1
o Min RAMS Buffer Fill Requirement (32 bits): Optional TLV element
that denotes the minimum milliseconds of data that RTP_Rx desires
to have in its buffer before allowing the data to be consumed by
the application.
RTP_Rx can have knowledge of its buffering requirements. These
requirements can be application and/or device specific. For
instance, RTP_Rx might need to have a certain amount of data in
its application buffer to handle transmission jitter and/or to be
able to support error-control methods. If BRS is told the minimum
buffering requirement of the receiver, BRS can tailor the burst(s)
more precisely, e.g., by choosing an appropriate starting point.
The methods used by RTP_Rx to determine this value are application
specific, and thus, out of the scope of this document.
If specified, the amount of backfill that will be provided by the
unicast bursts and any payload-specific information MUST NOT be
smaller than the specified value. Otherwise, the backfill will
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not be able to build up the desired level of buffer at RTP_Rx and
can cause buffer underruns.
Type: 2
o Max RAMS Buffer Fill Requirement (32 bits): Optional TLV element
that denotes the maximum milliseconds of data that RTP_Rx can
buffer without losing the data due to buffer overflow.
RTP_Rx can have knowledge of its buffering requirements. These
requirements can be application or device specific. For instance,
one particular RTP_Rx might have more physical memory than another
RTP_Rx, and thus, can buffer more data. If BRS knows the
buffering ability of the receiver, BRS can tailor the burst(s)
more precisely. The methods used by the receiver to determine
this value are application specific, and thus, out of scope.
If specified, the amount of backfill that will be provided by the
unicast bursts and any payload-specific information MUST NOT be
larger than this value. Otherwise, the backfill may cause buffer
overflows at RTP_Rx.
Type: 3
o Max Receive Bitrate (64 bits): Optional TLV element that denotes
the maximum bitrate (in bits per second) at which the RTP_Rx can
process the aggregation of the unicast burst(s) and any payload-
specific information that will be provided by BRS. The limits can
include local receiver limits as well as network limits that are
known to the receiver.
If specified, the total bitrate of the unicast burst(s) plus any
payload-specific information MUST NOT be larger than this value.
Otherwise, congestion and packet loss may occur.
Type: 4
o Request for Preamble Only (0 bits): Optional TLV element that
indicates that RTP_Rx is only requesting the preamble information
for the desired primary multicast stream(s). If this TLV element
exists in the RAMS-R message, BRS MUST NOT send any burst packets
other than the preamble packets. Since this TLV element does not
carry a Value field, the Length field MUST be set to zero.
Type: 5
The semantics of the RAMS-R feedback message is independent of the
payload type.
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7.3. RAMS Information
The RAMS Information message is identified by SFMT=2. This message
is used to describe the unicast burst that will be sent for rapid
acquisition. It also includes other useful information for RTP_Rx as
described below.
A separate RAMS-I message with the appropriate response code is sent
in the unicast burst RTP session by BRS for each primary multicast
stream that has been requested by RTP_Rx. In each of these RAMS-I
messages, the media sender SSRC and packet sender SSRC fields are
both set to the SSRC of BRS, which equals the SSRC of the respective
primary multicast stream.
The FCI field MUST contain only one RAMS Information. The FCI field
has the structure depicted in Figure 8.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SFMT=2 | MSN | Response |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: Optional TLV-encoded Fields (and Padding, if needed) :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 8: FCI field syntax for the RAMS Information message
A RAMS-I message has the following fields:
o Message Sequence Number (8 bits) : Mandatory field that denotes
the sequence number of the RAMS-I message for the particular media
sender SSRC specified in the message header. The MSN value SHALL
be set to zero only when a new RAMS request is received. During
rapid acquisition, the same RAMS-I message MAY be repeated for
redundancy purposes without incrementing the MSN value. If an
updated RAMS-I message will be sent (either with a new information
or an updated information), the MSN value SHALL be incremented by
one. In the MSN field, the regular wrapping rules apply.
o Response (16 bits): Mandatory field that denotes the response
code for this RAMS-I message. This document defines several
initial response codes and registers them with IANA. If a new
vendor-neutral response code will be defined, it MUST be
registered with IANA through the guidelines specified in
Section 11.6. If the new response code is intended to be used
privately by a vendor, there is no need for IANA management.
Instead, the vendor MUST use the private extension mechanism
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(Section 7.1.2) to convey its message and MUST indicate this by
putting zero in the Response field.
The following TLV elements have been defined for the RAMS-I messages:
o Media Sender SSRC (32 bits): Optional TLV element that specifies
the media sender SSRC of the unicast burst stream. While this
information is already available in the message header, it can be
useful to repeat it in an explicit field. If FT_Ap that received
the RAMS-R message is associated with a single primary multicast
stream but the requested media sender SSRC does not match the SSRC
of the RTP stream associated with this FT_Ap, BRS includes this
TLV element in the initial RAMS-I message to let RTP_Rx know that
the media sender SSRC has changed. If the two SSRCs match, there
is no need to include this TLV element.
Type: 31
o RTP Seqnum of the First Packet (16 bits): TLV element that
specifies the RTP sequence number of the first packet that will be
sent in the respective RTP stream. This allows RTP_Rx to know
whether one or more packets sent by BRS have been dropped at the
beginning of the stream. If BRS accepts the RAMS request, this
element exists. If BRS rejects the RAMS request, this element
SHALL NOT exist.
Type: 32
o Earliest Multicast Join Time (32 bits): TLV element that
specifies the delta time (in ms) between the arrival of the first
RTP packet in the RTP stream (which could be a burst packet or a
payload-specific packet) and the earliest time instant when RTP_Rx
sends an SFGMP Join message to join the multicast session. A zero
value in this field means that RTP_Rx can send the SFGMP Join
message right away.
If the RAMS request has been accepted, BRS sends this field at
least once, so that RTP_Rx knows when to join the multicast
session. If the burst request has been rejected as indicated in
the Response field, this field MUST be set to zero. In that case,
it is up to RTP_Rx when or whether to join the multicast session.
When BRS serves two or more bursts and sends a separate RAMS-I
message for each burst, the join times specified in these RAMS-I
messages should correspond to more or less the same time instant,
and RTP_Rx sends the SFGMP Join message based on the earliest join
time.
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Type: 33
o Burst Duration (32 bits): Optional TLV element that denotes the
duration of the burst, i.e., the delta difference between the
first and the last burst packet, that BRS is planning to send (in
ms) in the respective RTP stream. In the absence of additional
stimulus, BRS will send a burst of this duration. However, the
burst duration can be modified by subsequent events, including
changes in the primary multicast stream and reception of RAMS-T
messages.
BRS MUST terminate the flow in a deterministic timeframe, even if
it does not get a RAMS-T or a BYE from RTP_Rx. It is OPTIONAL to
send this field in a RAMS-I message when the burst request is
accepted. If the burst request has been rejected as indicated in
the Response field, this field MAY be omitted or set to zero.
Type: 34
o Max Transmit Bitrate (64 bits): Optional TLV element that denotes
the maximum bitrate (in bits per second) that will be used by BRS
for the RTP stream associated with this RAMS-I message.
Type: 35
The semantics of the RAMS-I feedback message is independent of the
payload type.
7.4. RAMS Termination
The RAMS Termination message is identified by SFMT=3.
The RAMS Termination is used to assist BRS in determining when to
stop the burst. A separate RAMS-T message is sent in the unicast
burst RTP session by RTP_Rx for each primary multicast stream that
has been served by BRS. Each of these RAMS-T messages has the
appropriate media sender SSRC populated in its message header.
If RTP_Rx wants BRS to stop a burst prematurely, it sends a plain
RAMS-T message as described below. Upon receiving this message, BRS
stops the respective burst immediately. If RTP_Rx wants BRS to
terminate all of the bursts, it needs to send all of the respective
RAMS-T messages in a single compound RTCP packet.
The default behavior for RTP_Rx is to send a RAMS-T message
immediately after it joined the multicast session and started
receiving multicast packets. In that case, RTP_Rx includes the
sequence number of the first RTP packet received in the primary
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multicast stream in the RAMS-T message. With this information, BRS
can decide when to terminate the unicast burst.
The FCI field MUST contain only one RAMS Termination. The FCI field
has the structure depicted in Figure 9.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SFMT=3 | Reserved |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: Optional TLV-encoded Fields (and Padding, if needed) :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 9: FCI field syntax for the RAMS Termination message
o Extended RTP Seqnum of First Multicast Packet (32 bits): Optional
TLV element that specifies the extended RTP sequence number of the
first packet received from the SSM session for a particular
primary multicast stream. The low 16 bits contain the sequence
number of the first packet received from the SSM session, and the
most significant 16 bits extend that sequence number with the
corresponding count of sequence number cycles, which can be
maintained according to the algorithm in Appendix A.1 of
[RFC3550].
Type: 61
The semantics of the RAMS-T feedback message is independent of the
payload type.
8. SDP Signaling
8.1. Definitions
The syntax of the 'rtcp-fb' attribute has been defined in [RFC4585].
Here we add the following syntax to the 'rtcp-fb' attribute (the
feedback type and optional parameters are all case sensitive):
(In the following ABNF [RFC5234], fmt, SP and CRLF are used as
defined in [RFC4566].)
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rtcp-fb-syntax = "a=rtcp-fb:" rtcp-fb-pt SP rtcp-fb-val CRLF
rtcp-fb-pt = "*" ; wildcard: applies to all formats
/ fmt ; as defined in SDP spec
rtcp-fb-val = "nack" SP "rai"
The following parameter is defined in this document for use with
'nack':
o 'rai' stands for Rapid Acquisition Indication and indicates the
use of RAMS messages as defined in Section 7.
This document also defines a new media-level SDP attribute ('rams-
updates') that indicates whether BRS supports updated request
messages or not. This attribute is used in a declarative manner and
no Offer/Answer Model behavior is specified. If BRS supports updated
request messages and this attribute is included in the SDP
description, RTP_Rx can send updated requests. BRS might or might
not be able to accept value changes in every field in an updated
RAMS-R message. However, if the 'rams-updates' attribute is not
included in the SDP description, RTP_Rx SHALL NOT send updated
requests. RTP_Rx MAY still repeat its initial request without
changes, though.
8.2. Requirements
The use of SDP to describe the RAMS entities normatively requires the
support for:
o The SDP grouping framework and flow identification (FID) semantics
[RFC5888]
o The RTP/AVPF profile [RFC4585]
o The RTP retransmission payload format [RFC4588]
o The RTCP extensions for SSM sessions with unicast feedback
[RFC5760]
o The multicast RTCP port attribute [I-D.ietf-avt-rtcp-port-for-ssm]
o Multiplexing RTP and RTCP on a single port on both endpoints in
the unicast session[RFC5761]
The support for the source-specific media attributes [RFC5576] can
also be needed.
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8.3. Example and Discussion
This section provides a declarative SDP [RFC4566] example for
enabling rapid acquisition of multicast RTP sessions.
v=0
o=ali 1122334455 1122334466 IN IP4 rams.example.com
s=Rapid Acquisition Example
t=0 0
a=group:FID 1 2
a=rtcp-unicast:rsi
m=video 41000 RTP/AVPF 98
i=Primary Multicast Stream
c=IN IP4 233.252.0.2/255
a=source-filter:incl IN IP4 233.252.0.2 198.51.100.1
a=rtpmap:98 MP2T/90000
a=multicast-rtcp:42000
a=rtcp:43000 IN IP4 192.0.2.1
a=rtcp-fb:98 nack
a=rtcp-fb:98 nack rai
a=ssrc:123321 cname:iptv-ch32@rams.example.com
a=rams-updates
a=mid:1
m=video 51000 RTP/AVPF 99
i=Unicast Retransmission Stream (Ret. and Rapid Acq. Support)
c=IN IP4 192.0.2.1
a=sendonly
a=rtpmap:99 rtx/90000
a=rtcp-mux
a=fmtp:99 apt=98;rtx-time=5000
a=mid:2
Figure 10: Example SDP for a single-channel RAMS scenario
In this example SDP description, we have a primary multicast (source)
stream and a unicast retransmission stream. The source stream is
multicast from a distribution source (with a source IP address of
198.51.100.1) to the multicast destination address of 233.252.0.2 and
port 41000. The corresponding RTCP traffic is multicast to the same
multicast destination address at port 42000. Here, we are assuming
that the multicast RTP and RTCP ports are carefully chosen so that
different RTP and RTCP streams do not collide with each other.
The feedback target (FT_Ap) residing on RS (with an address of
192.0.2.1) at port 43000 is declared with the "a=rtcp" line
[RFC3605]. The support for the conventional retransmission is
indicated through the "a=rtcp-fb:98 nack" line. The RTP receiver(s)
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can report missing packets on the source stream to the feedback
target and request retransmissions. The SDP above includes the
"a=sendonly" line for the media description of the retransmission
stream since the retransmissions are sent in only one direction.
The support for rapid acquisition is indicated through the "a=rtcp-
fb:98 nack rai" line. The parameter 'rtx-time' applies to both the
conventional retransmissions and rapid acquisition. However, when
rapid acquisition is enabled, the standard definition for the
parameter 'rtx-time' given in [RFC4588] is not followed. Instead,
when rapid acquisition support is enabled, 'rtx-time' specifies the
time in milliseconds that BRS keeps an RTP packet in its cache
available for retransmission (measured from the time the packet was
received by BRS, not from the time indicated in the packet
timestamp).
Once an RTP_Rx has acquired an SDP description, it can ask for rapid
acquisition before it joins a primary multicast RTP session. To do
so, it sends a RAMS-R message to the feedback target of that primary
multicast RTP session. If FT_Ap is associated with only one RTP
stream, RTP_Rx does not need to learn the SSRC of that stream via an
out-of-band method. If BRS accepts the rapid acquisition request, it
will send an RAMS-I message with the correct SSRC identifier. If
FT_Ap is associated with a multi-stream RTP session and RTP_Rx is
willing to request rapid acquisition for the entire session, RTP_Rx
again does not need to learn the SSRCs via an out-of-band method.
However, if RTP_Rx intends to request a particular subset of the
primary multicast streams, it must learn their SSRC identifiers and
list them in the RAMS-R message. Since this RTP_Rx has not yet
received any RTP packets for the primary multicast stream(s), RTP_Rx
must in this case learn the SSRC value(s) from the 'ssrc' attribute
of the media description [RFC5576]. In addition to the SSRC value,
the 'cname' source attribute must also be present in the SDP
description [RFC5576].
Listing the SSRC values for the primary multicast streams in the SDP
file does not create a problem in SSM sessions when an SSRC collision
occurs. This is because in SSM sessions, an RTP_Rx that observed an
SSRC collision with a media sender must change its own SSRC [RFC5760]
by following the rules defined in [RFC3550].
A feedback target that receives a RAMS-R feedback message becomes
aware that the prediction chain at RTP_Rx side has been broken or
does not exist anymore. If the necessary conditions are satisfied
(as outlined in Section 7 of [RFC4585]) and available resources
exist, BRS can react to the RAMS-R message by sending any transport-
layer (and optional payload-specific, when allowed) feedback
message(s) and starting the unicast burst.
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In this section, we considered the simplest scenario where the
primary multicast RTP session carried only one stream and RTP_Rx
wanted to rapidly acquire this stream only. Best practices for
scenarios where the primary multicast RTP session carries two or more
streams or RTP_Rx wants to acquire one or more streams from multiple
primary multicast RTP sessions at the same time are presented in
[I-D.begen-avt-rams-scenarios].
9. NAT Considerations
For a variety of reasons, one or more NAPT devices (hereafter simply
called NAT) can exist between RTP_Rx and RS. NATs have a variety of
operating characteristics for UDP traffic [RFC4787]. For a NAT to
permit traffic from BRS to arrive at RTP_Rx, the NAT(s) must first
either:
a. See UDP traffic sent from RTP_Rx (which is on the 'inside' of the
NAT) to BRS (which is on the 'outside' of the NAT). This traffic
has the same transport address as the subsequent response
traffic, or;
b. Be configured to forward certain ports (e.g., using HTML
configuration, UPnP IGD [UPnP-IGD], DLNA [DLNA]). Details of
this are out of scope of this document.
For both (a) and (b), RTP_Rx is responsible for maintaining the NAT's
state if it wants to receive traffic from the BRS on that port. For
(a), RTP_Rx MUST send UDP traffic to keep the NAT binding alive, at
least every 30 seconds [RFC4787]. While (a) is more like an
automatic/dynamic configuration, (b) is more like a manual/static
configuration.
When RTP_Rx sends a request (RAMS-R) message to FT as unicast
feedback in the primary multicast RTP session, and the request is
received by FT, a new unicast burst RTP session will be established
between BRS and RTP_Rx.
While the FT and BRS ports on RS are already signaled via out-of-band
means (e.g., SDP), RTP_Rx needs to convey the RTP and RTCP ports it
wants to use on its side for the new session. Since there are two
RTP sessions (one multicast and one unicast) involved during this
process and one of them is established upon a feedback message sent
in the other one, this requires an explicit port mapping method.
Applications using RAMS MUST support the solution presented in
[I-D.ietf-avt-ports-for-ucast-mcast-rtp] both on the RS and RTP_Rx
side to allow RTP receivers to use their desired ports and to support
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RAMS behind NAT devices. The port mapping message exchange needs to
take place whenever RTP_Rx intends to make use of the RAMS protocol
for rapidly acquiring a specific multicast RTP session, prior to
joining the associated SSM session.
10. Security Considerations
Applications that are using RAMS make heavy use of the unicast
feedback mechanism described in [RFC5760], the payload format defined
in [RFC4588] and the port mapping solution specified in
[I-D.ietf-avt-ports-for-ucast-mcast-rtp]. Thus, these applications
are subject to the general security considerations discussed in
[RFC5760], [RFC4588] and [I-D.ietf-avt-ports-for-ucast-mcast-rtp].
In this section, we give an overview of the guidelines and
suggestions described in these specifications from a RAMS
perspective. We also discuss the security considerations that
explicitly apply to applications using RAMS.
First of all, much of the session description information is
available in the SDP descriptions that are distributed to the media
senders, retransmission servers and RTP receivers. Adequate security
measures are RECOMMENDED to ensure the integrity and authenticity of
the SDP descriptions so that transport addresses of the media
senders, distribution sources, feedback targets as well as other
session-specific information can be protected.
Compared to an RTCP NACK message that triggers one or more
retransmissions, a RAMS Request (RAMS-R) message can trigger a new
burst stream to be sent by the retransmission server. Depending on
the application-specific requirements and conditions existing at the
time of the RAMS-R reception by the retransmission server, the
resulting burst stream can potentially contain a large number of
retransmission packets. Since these packets are sent at a faster
than the nominal rate, RAMS consumes more resources on the
retransmission servers, RTP receivers and the network. In
particular, this can make denial-of-service attacks more intense, and
hence, more harmful than attacks that target ordinary retransmission
sessions.
Following the suggestions given in [RFC4588], counter-measures SHOULD
be taken to prevent tampered or spoofed RTCP packets. Tampered
RAMS-R messages can trigger inappropriate burst streams or alter the
existing burst streams in an inappropriate way. For example, if the
Max Receive Bitrate field is altered by a tampered RAMS-R message,
the updated burst can overflow the buffer at the receiver side, or
oppositely, can slow down the burst to the point that it becomes
useless. Tampered RAMS Termination (RAMS-T) messages can terminate
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valid burst streams prematurely resulting in gaps in the received RTP
packets. RAMS Information (RAMS-I) messages contain fields that are
critical for a successful rapid acquisition. Any tampered
information in the RAMS-I message can easily cause an RTP receiver to
make wrong decisions. Consequently, the RAMS operation can fail.
While most of the denial-of-service attacks can be prevented by the
integrity and authenticity checks enabled by Secure RTP (SRTP)
[RFC3711], an attack can still be started by legitimate endpoints
that send several valid RAMS-R messages to a particular feedback
target in a synchronized fashion and very short amount of time.
Since a RAMS operation can temporarily consume a large amount of
resources, a series of the RAMS-R messages can temporarily overload
the retransmission server. In these circumstances, the
retransmission server can, for example, reject incoming RAMS requests
until its resources become available again. One means to ameliorate
this threat is to apply a per-endpoint policing mechanism on the
incoming RAMS requests. A reasonable policing mechanism should
consider application-specific requirements and minimize false
negatives.
In addition to the denial-of-service attacks, man-in-the-middle and
replay attacks can also be harmful. However, RAMS itself does not
bring any new risks or threats other than the ones discussed in
[RFC5760].
[RFC4588] RECOMMENDS that the cryptography mechanisms are used for
the retransmission payload format to provide protection against known
plain-text attacks. As discussed in [RFC4588], the retransmission
payload format sets the timestamp field in the RTP header to the
media timestamp of the original packet and this does not compromise
the confidentiality. Furthermore, if cryptography is used to provide
security services on the original stream, then the same services,
with equivalent cryptographic strength, MUST be provided on the
retransmission stream per [RFC4588].
To protect the RTCP messages from man-in-the-middle and replay
attacks, the RTP receivers and retransmission server SHOULD perform a
DTLS-SRTP handshake [RFC5764] over the RTCP channel. Because there
is no integrity-protected signaling channel between an RTP receiver
and the retransmission server, the retransmission server MUST
maintain a list of certificates owned by legitimate RTP receivers, or
their certificates MUST be signed by a trusted Certificate Authority.
Once the DTLS-SRTP security is established, non-SRTCP-protected
messages received from a particular RTP receiver are ignored by the
retransmission server. To reduce the impact of DTLS-SRTP overhead
when communicating with different feedback targets on the same
retransmission server, it is RECOMMENDED that RTP receivers and the
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retransmission server both support TLS Session Resumption without
Server-side State [RFC5077]. To help scale SRTP to handle many RTP
receivers asking for retransmissions of identical data, implementors
may consider using the same SRTP key for SRTP data sent to the
receivers [I-D.ietf-avt-srtp-ekt] and consider the security of such
SRTP key sharing.
11. IANA Considerations
The following contact information shall be used for all registrations
in this document:
Ali Begen
abegen@cisco.com
Note to the RFC Editor: In the following, please replace "XXXX" with
the number of this document prior to publication as an RFC.
11.1. Registration of SDP Attributes
This document registers a new attribute name in SDP.
SDP Attribute ("att-field"):
Attribute name: rams-updates
Long form: Support for Updated RAMS Request Messages
Type of name: att-field
Type of attribute: Media level
Subject to charset: No
Purpose: See this document
Reference: [RFCXXXX]
Values: None
11.2. Registration of SDP Attribute Values
This document registers a new value for the 'nack' attribute to be
used with the 'rtcp-fb' attribute in SDP. For more information about
the 'rtcp-fb' attribute, refer to Sections 4.2 and 6.2 of [RFC4585].
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Value name: rai
Long name: Rapid Acquisition Indication
Usable with: nack
Reference: [RFCXXXX]
11.3. Registration of FMT Values
Within the RTPFB range, the following format (FMT) value is
registered:
Name: RAMS
Long name: Rapid Acquisition of Multicast Sessions
Value: 6
Reference: [RFCXXXX]
11.4. SFMT Values for RAMS Messages Registry
This document creates a new sub-registry for the sub-feedback message
type (SFMT) values to be used with the FMT value registered for RAMS
messages. The registry is called the SFMT Values for RAMS Messages
Registry. This registry is to be managed by the IANA according to
the Specification Required policy of [RFC5226].
The length of the SFMT field in the RAMS messages is a single octet,
allowing 256 values. The registry is initialized with the following
entries:
Value Name Reference
----- -------------------------------------------------- -------------
0 Reserved [RFCXXXX]
1 RAMS Request [RFCXXXX]
2 RAMS Information [RFCXXXX]
3 RAMS Termination [RFCXXXX]
4-254 Assignable - Specification Required
255 Reserved [RFCXXXX]
The SFMT values 0 and 255 are reserved for future use.
Any registration for an unassigned SFMT value needs to contain the
following information:
o Contact information of the one doing the registration, including
at least name, address, and email.
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o A detailed description of what the new SFMT represents and how it
shall be interpreted.
New RAMS functionality is intended to be introduced by using the
extension mechanism within the existing RAMS message types not by
introducing new message types unless it is absolutely necessary.
11.5. RAMS TLV Space Registry
This document creates a new IANA TLV space registry for the RAMS
extensions. The registry is called the RAMS TLV Space Registry.
This registry is to be managed by the IANA according to the
Specification Required policy of [RFC5226].
The length of the Type field in the TLV elements is a single octet,
allowing 256 values. The Type values 0 and 255 are reserved for
future use. The Type values between (and including) 128 and 254 are
reserved for private extensions.
The registry is initialized with the following entries:
Type Description Reference
---- -------------------------------------------------- -------------
0 Reserved [RFCXXXX]
1 Requested Media Sender SSRC(s) [RFCXXXX]
2 Min RAMS Buffer Fill Requirement [RFCXXXX]
3 Max RAMS Buffer Fill Requirement [RFCXXXX]
4 Max Receive Bitrate [RFCXXXX]
5 Request for Preamble Only [RFCXXXX]
6-30 Assignable - Specification Required
31 Media Sender SSRC [RFCXXXX]
32 RTP Seqnum of the First Packet [RFCXXXX]
33 Earliest Multicast Join Time [RFCXXXX]
34 Burst Duration [RFCXXXX]
35 Max Transmit Bitrate [RFCXXXX]
36-60 Assignable - Specification Required
61 Extended RTP Seqnum of First Multicast Packet [RFCXXXX]
62-127 Assignable - Specification Required
128-254 No IANA Maintenance
255 Reserved [RFCXXXX]
Any registration for an unassigned Type value needs to contain the
following information:
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o Contact information of the one doing the registration, including
at least name, address, and email.
o A detailed description of what the new TLV element represents and
how it shall be interpreted.
11.6. RAMS Response Code Space Registry
This document creates a new IANA TLV space registry for the RAMS
response codes. The registry is called the RAMS Response Code Space
Registry. This registry is to be managed by the IANA according to
the Specification Required policy of [RFC5226].
The length of the Response field is two octets, allowing 65536 codes.
However, the response codes have been classified and registered
following an HTTP-style code numbering in this document. New
response codes should be classified following the guidelines below:
Code Level Purpose
---------- ---------------
1xx Informational
2xx Success
3xx Redirection
4xx RTP Receiver Error
5xx Retransmission Server Error
The Response code 65536 is reserved for future use.
The registry is initialized with the following entries:
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Code Description Reference
----- -------------------------------------------------- -------------
0 A private response code is included in the message [RFCXXXX]
100 Parameter update for RAMS session [RFCXXXX]
200 RAMS request has been accepted [RFCXXXX]
201 Unicast burst has been completed [RFCXXXX]
400 Invalid RAMS-R message syntax
401 Invalid min buffer requirement in RAMS-R message [RFCXXXX]
402 Invalid max buffer requirement in RAMS-R message [RFCXXXX]
403 Invalid max bitrate requirement in RAMS-R message [RFCXXXX]
500 An unspecified BRS internal error has occurred [RFCXXXX]
501 BRS has insufficient bandwidth to start RAMS
session [RFCXXXX]
502 Burst is terminated due to network congestion [RFCXXXX]
503 BRS has insufficient CPU cycles to start RAMS
session [RFCXXXX]
504 RAMS functionality is not available on BRS [RFCXXXX]
505 RAMS functionality is not available for RTP_Rx [RFCXXXX]
506 RAMS functionality is not available for
the requested multicast stream [RFCXXXX]
507 BRS has no valid starting point available for
the requested multicast stream [RFCXXXX]
508 BRS has no reference information available for
the requested multicast stream [RFCXXXX]
509 BRS has no RTP stream matching the requested SSRC [RFCXXXX]
510 RAMS request to acquire the entire session
has been denied [RFCXXXX]
511 Only the preamble information is sent [RFCXXXX]
512 RAMS request has been denied due to a policy [RFCXXXX]
Any registration for an unassigned Response code needs to contain the
following information:
o Contact information of the one doing the registration, including
at least name, address, and email.
o A detailed description of what the new Response code describes and
how it shall be interpreted.
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12. Contributors
Dave Oran, Magnus Westerlund and Colin Perkins have contributed
significantly to this specification by providing text and solutions
to some of the issues raised during the development of this
specification.
13. Acknowledgments
The following individuals have reviewed the earlier versions of this
specification and provided helpful comments: Colin Perkins, Joerg
Ott, Roni Even, Dan Wing, Tony Faustini, Peilin Yang, Jeff Goldberg,
Muriel Deschanel, Orit Levin, Guy Hirson, Tom Taylor, Xavier Marjou,
Ye-Kui Wang, Zixuan Zou, Ingemar Johansson, Haibin Song, Ning Zong,
Jonathan Lennox, Jose Rey, Sean Sheedy and Keith Drage.
14. References
14.1. Normative References
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3376] Cain, B., Deering, S., Kouvelas, I., Fenner, B., and A.
Thyagarajan, "Internet Group Management Protocol, Version
3", RFC 3376, October 2002.
[RFC3810] Vida, R. and L. Costa, "Multicast Listener Discovery
Version 2 (MLDv2) for IPv6", RFC 3810, June 2004.
[RFC4604] Holbrook, H., Cain, B., and B. Haberman, "Using Internet
Group Management Protocol Version 3 (IGMPv3) and Multicast
Listener Discovery Protocol Version 2 (MLDv2) for Source-
Specific Multicast", RFC 4604, August 2006.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description
Protocol (SDP) Grouping Framework", RFC 5888, June 2010.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
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"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
July 2006.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
July 2006.
[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.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, June 2009.
[RFC3605] Huitema, C., "Real Time Control Protocol (RTCP) attribute
in Session Description Protocol (SDP)", RFC 3605,
October 2003.
[RFC5234] Crocker, D. and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF", STD 68, RFC 5234, January 2008.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009.
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
Header Extensions", RFC 5285, July 2008.
[I-D.ietf-avt-rapid-rtp-sync]
Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
Flows", draft-ietf-avt-rapid-rtp-sync-11 (work in
progress), May 2010.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761, April 2010.
[I-D.ietf-avt-rtcp-port-for-ssm]
Begen, A., "RTP Control Protocol (RTCP) Port for Multicast
Sessions", draft-ietf-avt-rtcp-port-for-ssm-00 (work in
progress), June 2010.
[I-D.ietf-avt-ports-for-ucast-mcast-rtp]
Begen, A. and B. Steeg, "Port Mapping Between Unicast and
Multicast RTP Sessions",
draft-ietf-avt-ports-for-ucast-mcast-rtp-02 (work in
progress), May 2010.
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[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764, May 2010.
[RFC5077] Salowey, J., Zhou, H., Eronen, P., and H. Tschofenig,
"Transport Layer Security (TLS) Session Resumption without
Server-Side State", RFC 5077, January 2008.
[RFC5226] Narten, T. and H. Alvestrand, "Guidelines for Writing an
IANA Considerations Section in RFCs", BCP 26, RFC 5226,
May 2008.
14.2. Informative References
[RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768,
August 1980.
[I-D.begen-avt-rams-scenarios]
Begen, A., "Considerations for RAMS Scenarios",
draft-begen-avt-rams-scenarios-00 (work in progress),
October 2009.
[I-D.ietf-avt-rtp-cnames]
Begen, A., Perkins, C., and D. Wing, "Guidelines for
Choosing RTP Control Protocol (RTCP) Canonical Names
(CNAMEs)", draft-ietf-avt-rtp-cnames-00 (work in
progress), June 2010.
[I-D.ietf-avt-multicast-acq-rtcp-xr]
Begen, A. and E. Friedrich, "Multicast Acquisition Report
Block Type for RTP Control Protocol (RTCP) Extended
Reports (XRs)", draft-ietf-avt-multicast-acq-rtcp-xr-01
(work in progress), May 2010.
[I-D.ietf-avt-ecn-for-rtp]
Westerlund, M., Johansson, I., Perkins, C., and K.
Carlberg, "Explicit Congestion Notification (ECN) for RTP
over UDP", draft-ietf-avt-ecn-for-rtp-01 (work in
progress), March 2010.
[I-D.ietf-fecframe-interleaved-fec-scheme]
Begen, A., "RTP Payload Format for 1-D Interleaved Parity
FEC", draft-ietf-fecframe-interleaved-fec-scheme-09 (work
in progress), January 2010.
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[RFC4787] Audet, F. and C. Jennings, "Network Address Translation
(NAT) Behavioral Requirements for Unicast UDP", BCP 127,
RFC 4787, January 2007.
[RFC5762] Perkins, C., "RTP and the Datagram Congestion Control
Protocol (DCCP)", RFC 5762, April 2010.
[I-D.ietf-avt-srtp-ekt]
McGrew, D., Andreasen, F., Wing, D., and L. Dondeti,
"Encrypted Key Transport for Secure RTP",
draft-ietf-avt-srtp-ekt-00 (work in progress), March 2010.
[UPnP-IGD]
Forum, UPnP., "Universal Plug and Play (UPnP) Internet
Gateway Device (IGD)", November 2001.
[DLNA] , DLNA., "http://www.dlna.org/home".
[IC2009] Begen, A., Glazebrook, N., and W. VerSteeg, "Reducing
Channel Change Times in IPTV with Real-Time Transport
Protocol (IEEE Internet Computing)", May 2009.
Authors' Addresses
Bill VerSteeg
Cisco
5030 Sugarloaf Parkway
Lawrenceville, GA 30044
USA
Email: billvs@cisco.com
Ali Begen
Cisco
181 Bay Street
Toronto, ON M5J 2T3
Canada
Email: abegen@cisco.com
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Tom VanCaenegem
Alcatel-Lucent
Copernicuslaan 50
Antwerpen, 2018
Belgium
Email: Tom.Van_Caenegem@alcatel-lucent.be
Zeev Vax
Microsoft Corporation
1065 La Avenida
Mountain View, CA 94043
USA
Email: zeevvax@microsoft.com
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