One document matched: draft-wu-avtcore-multiplex-multisource-endpoint-00.xml
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docName="draft-wu-avtcore-multiplex-multisource-endpoint-00.txt"
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<front>
<title abbrev="Bandwidth and RTCP Timing">Bandwidth and RTCP timing issues
for multi-source endpoint</title>
<author fullname="Qin Wu" initials="Q." surname="Wu">
<organization>Huawei</organization>
<address>
<postal>
<street>101 Software Avenue, Yuhua District</street>
<city>Nanjing</city>
<region>Jiangsu</region>
<code>210012</code>
<country>China</country>
</postal>
<email>sunseawq@huawei.com</email>
</address>
</author>
<date year="2012" />
<area>Real-time Applications and Infrastructure Area</area>
<workgroup>Audio/Video Transport Working Group</workgroup>
<keyword>RFC</keyword>
<keyword>Request for Comments</keyword>
<keyword>I-D</keyword>
<keyword>Internet-Draft</keyword>
<keyword>Real Time Control Protocol</keyword>
<abstract>
<t>This document discusses bandwidth issues and RTCP timing rule issues
that arise when the multi-source endpoint multiplexing all the media
type in one RTP session and follows RFC3550 timing rules. It provides
recommendations for multi-source host sending multiple media types in
the same session.</t>
</abstract>
</front>
<middle>
<section anchor="intro" title="Introduction">
<t>Multiplexing is a method by which multiple streams are combined into
one stream over a shared medium. In RTP, multiplexing is provided by the
destination transport address (network address and port number) which is
different for each RTP session. Given each media type having different
requirement for bandwidth, multiple media types (e.g., Separate audio
and video streams) are usually not carried in a single RTP session. </t>
<t>However for some applications that use unicast transport, e.g., in
RTCWeb application, multiple-source hosts may use a different SSRC for
each medium but sending them in the same RTP session, which reduces
communication failure due to NAT and firewall when using multiple RTP
sessions or multiple transport flow. If these multi-source hosts still
follow RFC3550 timing rules, audio and video are multiplexed onto a
single RTP session and share a common session bandwidth, the audio flows
sending at much lower rates will waste a large amount of bandwidth.</t>
<t>This document provides recommendations for multi-source host sending
multiple media types in the same session.</t>
</section>
<section title="Terminology">
<section title="Standards Language">
<t>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 <xref
target="RFC2119">RFC 2119</xref>.<list style="hanging">
<t hangText="Multi-Source Endpoint"><vspace blankLines="1" />End
system with multiple sources generated from one host and running
on that host. One example of multi-source endpoint is an RTP
endpoint which has multiple capture devices of the same media type
and characteristics.</t>
<t hangText="Single-Source Endpoint"><vspace blankLines="0" />End
system with single source generated from one host described in
RFC3550.</t>
</list></t>
</section>
</section>
<section title="Use Cases for multi-source host communications">
<section title="One to one Communication between multi-source endpoints">
<figure title="Figure 1">
<artwork>
Multiple-Source Multiple-Source
Endpoint1 Endpoint2
+--------+ +--------+
| | Common Transport | |
| | | |
| |Transport flow1(v1,a1) |
SSRC-V1---| |------------------>| |--- SSRC-V2
| | | |
| | | |
| | | |
| | | |
SSRC-A1---| |Transport flow1(v2,a2) |
| |<------------------| +----SSRC-A2
| | | |
| | | |
| | | |
+--------+ +--------+
</artwork>
</figure>
<t>In the figure 1, one to one communication between multiple-source
endpoint1 and multiple-source endpoint2 takes place. In order to
reduce unicast transport flow to facilitate NAT/FW traversal,
Multiple-Source endpoint 1 and multiple-source endpoint2 in one to one
communication share the same transport. Both video stream v1 and audio
stream a1 from multiple-source endpoint 1 are multiplexed in the RTP
session1 and transmitted over the transport flow1 to multiple-source
endpoint 2. Similarly, both video stream v2 and audio stream a2 from
multiple-source endpoint2 are multiplexed in the RTP session 2 and
transmitted over the same transmitted over the same transport flow 1
as multiple source endpoint1.</t>
<t>Multiple-source endpoint1 sends RTCP SR and RR packets for every
active media stream it’s receiving (i.e.,v2,a2) from every local
source (i.e.,v1,a1), which wastes a lot of bandwidth for redundant
statistics reports. So does multiple-source endpoint2.</t>
<t>Also the bandwidth usage for each stream is limited by its media
data rate and audio stream usually consumes lower bandwidth than video
stream (e.g.,high quality audio with 80kbps and high quality video
with 350 kbps). When audio stream and video stream share the same
transport, i.e., sharing the common session bandwidth between audio
and video, audio stream will still be sent at its own media data rate
far below the common session bandwidth and therefore waste lots of
bandwidth provided by the transport. Further when reception reports
are used for audio stream and follows RFC3550 timing rule, RTCP SR and
RR packets for active audio stream will be reported more frequent than
when audio and video are carried in separate transports since more
RTCP bandwidth are allocated to audio stream reporting when more
session bandwidth are allocated to audio stream.</t>
</section>
<section title="Multi-party Communication between multi-source endpoints">
<figure title="Figure 2">
<artwork>
+---+
| |
Transport flow 1(v1,a1)-----------+
Transport flow 1(v2,a2) |
+--+ c +--------> |
| | e | | Multiple |-----SSRC-V2
| | n | | Source |
+-----------+ | | t | | Endpoint2 |
| <---| | e | | |
SSRC-V1---| | | r | | +-----SSRC-A2
| Multiple | | | +-----------+
| Source | | s |
| Endpoint1 | | e |
SSRC-A1---| <---| | r | +-----------+
| | | | v | | |
+-----------+ | | e | | |
| | r | | Multiple |-----SSRC-V3
| | Source |
Transport flow 2(v1,a1) Endpoint3 |
Transport flow 2(v3,a3) |
---| +--------> +-----SSRC-A3
| | +-----------+
| |
| |
+---+
</artwork>
</figure>
<t>In figure 2, multiple-source endpoint1 communicates with
multiple-source endpoint2 and multiple-source endpoint3 in the same
session. In order to reduce unicast transport flow to facilitate
NAT/FW traversal, multiple-source endpoint 1 shares the same transport
flow 1 with multiple-source endpoint2 to send video stream v1 and
audio stream a1 and receive video stream v2 and audio stream v2
simultaneously. Similarly, multiple-source endpoint 1 shares the same
transport flow 2 with multiple-source endpoint3 to send video stream
v1 and audio stream a1 and receive video stream v3 and audio stream a3
simultaneously.</t>
<t>Multiple-source endpoint1 sends RTCP SR and RR packets for every
active media stream it’s receiving (i.e.,v2,a2,v3,a3) from every local
source (i.e.,v1,a1), which wastes a lot of bandwidth for redundant
statistics reports. So does multiple-source endpoint2.</t>
<t> Also the bandwidth usage for each stream is limited by its media
data rate and audio stream usually consumes lower bandwidth than video
stream (e.g.,high quality audio with 80kbps and high quality video
with 350 kbps). When audio stream and video stream share the same
transport, i.e., sharing the common session bandwidth between audio
and video, audio stream will still be sent at its own media data rate
far below the common session bandwidth and therefore waste lots of
bandwidth provided by the transport. Further when reception reports
are used for audio stream and follows RFC3550 timing rule, RTCP SR and
RR packets for active audio stream will be reported more frequent than
when audio and video are carried in separate transports since more
RTCP bandwidth are allocated to audio stream reporting when more
session bandwidth are allocated to audio stream.</t>
</section>
<section title="Communication between multi-source endpoint multiplexing each medium in separate RTP session and multiple-source endpoint multiplexing all mediums in one RTP session">
<figure>
<artwork>
Multiple-Source Multiple-Source
Endpoint1 Endpoint2
+--------+ +--------+
| | Various Transports| |
| | | |
| |Transport flow1(v1,a1) |
SSRC-V1---| |------------------>| |--- SSRC-V2
| | | |
| |Transport flow2(v2)| |
| |<------------------| |
| | | |
SSRC-A1---| |Transport flow3(a2)| |
| |<------------------| +----SSRC-A2
| | | |
| | | |
| | | |
+--------+ +--------+
</artwork>
</figure>
<t>In the figure 3, multiple-source endpoint1 residing outside of the
firewall communicates with multiple-source endpoint2 residing inside
of the firewall. Unlike one to one communication described in section
3.1, multiple-source endpoint2 multiplexes each medium (i.e.,v2,a2) in
separate transports (i.e.,transport flow2,transport flow3)to
multiple-source endpoint1 while multiple-source endpoint1 multiplexes
all mediums(ie.,v1,a1) in one unicast transport(i.e.,transport
flow1)to multiple-source endpoint2.</t>
</section>
</section>
<section title="Discuss">
<t>The RTCP bandwidth fraction is derived from the media data rate. If
audio and video are sent as two separate RTP sessions, they would
naturally have different RTCP bandwidth fractions, since the two media
types have different rates.</t>
<t>However, if audio and video are multiplexed onto a single RTP
session, a common session bandwidth would have to be chosen. This common
bandwidth will likely be inappropriate for one of the media types,
leading to the situation where some media flows use their allocation,
while some flows send at a rate that is quite different to the session
bandwidth. In the common case, the video sends at the session bandwidth,
while the audio flows send at much lower rates. The RTCP bandwidth is
derived from the session bandwidth, which means it's appropriate for the
video, but is too high for the audio. In the worst case, the audio flows
can have more RTCP than video flows, which is very wasteful. Fixing this
is potentially difficult, since the session bandwidth concept is baked
into all the RTCP timing rules. </t>
</section>
<section title="Recommendations">
<t>Senders and receivers which are not multi-source endpoints are not
affected by bandwidth issues associated with multi-source endpoint and
should follow RFC3550 timing rules[RFC3550], no special accommodation is
required. </t>
<t>Senders and receivers which are multi-source endpoints and sending or
receiving multiple media types over different transport should be
treated in the same way as single source endpoints dealing with multiple
streams in the same media type.</t>
<t>Senders and receivers which are multi-source endpoints and sending or
receiving multiple media types MAY multiplex/demultiplex each media type
in separate RTP session or multiplex/demultiplex all media types in one
RTP session. The multiplexing/demultiplexing mode to be employed in two
directions between senders and receivers should be configurable. It is
RECOMMENDED that<list style="numbers">
<t>As the default behavior, Senders and receivers use the media
bundling mechanism [BUNDLE] in two directions between senders and
receivers, i.e., multiplexing/demulplexing separate media types in
one RTP session over the same lower layer transport. </t>
<t>Configuration or local policy on the senders or receivers can
override the default Mechanism specified in Option 1 above in one or
two direction. Therefore senders and receivers MAY be configured
with the different multiplexing mode. </t>
<t>Dynamic multiplexing negotiation mechanism [BUNDLE] can be used
to signal which multiplexing mechanism is used between senders and
receivers and override the default mechanism specified in Option 1
and 2 above. The employed multiplexing negotiation mechanism is
outside the scope of this document. </t>
</list></t>
<t>Multi-source endpoints multiplexing each media type in separate RTP
session SHOULD be treated as multiple endpoints for each media type.</t>
<section title="Choosing RTCP Bandwidth">
<t>Multi-source endpoint multiplexing multiple media types in the same
RTP session SHOULD specify RTCP bandwidth using SDP, in a “b=RS” line
or a “b=RR” line rather than choosing 5% of session bandwidth for RTCP
bandwidth, especially when audio stream and video stream are sent over
the same transport and the media data rate of audio stream is far less
than video stream.</t>
<t>RTCP bandwidth for video stream MAY still follow RFC3550
recommendation, i.e., the fraction of the session bandwidth added for
RTCP be fixed at 5%. </t>
<t>RTCP bandwidth for video = session bandwidth *5%</t>
<t>However given lower bandwidth usage of audio stream, RTCP bandwidth
for audio stream is RECOMMENDED to be specified using SDP in a “b=RS”
line or a “b=RR” line.</t>
<t>RTCP bandwidth for audio stream SHOULD be calculated based on the
following formula:</t>
<t>RTCP bandwidth for audio = ((audio codec maximum bitrate*20%)+
audio codec maximum bitrate)*5%</t>
<t>Here a 20% RTP packet overhead is added to the data rate to
calculate the required RTCP bandwidth.</t>
</section>
<section title="Maintaining the number of session members and senders">
<t>When multi-Source endpoints multiplexing all media types in the
same RTP session join the session, they SHOULD be counted as only one
active sender. To achieve this, the sender which is multi-source
endpoint SHOULD not send Reception reports (e.g., SR/RR RTCP packets)
from all local sources to each remote source in the receiving side,
Similarly the Receiver which is multi-source endpoint SHOULD not
receive reception reports(e.g., SR/RR RTCP packets from each remote
source in the sending side to all local sources. Instead, one
designated local source from the senders within multi-source endpoint
should be chosen as reporting source for other local sources within
the sender. </t>
<t>When multi-Source endpoints multiplexing each media type in
separated RTP session join the session, they SHOULD treated as
multiple single-source endpoint multiplexing for each media type and
SHOULD be counted as multiple active senders. The number of active
senders within multi-source endpoint is decided by the number of local
source which is sending reception reports. </t>
<t>For the number of session members, it depends on the number of
multiple-source endpoints E and the number of local source in each
multiple-source endpoint S. It can be estimated or calculated
according to the rules in RFC 3550 section 6.2.1, based on received
valid SSRC in the SDES packet or RTP packets.</t>
</section>
<section title="RTCP Reporting Interval calculation">
<t>The RTCP packet interval calculation SHOULD consider RTCP bandwidth
estimation and signaling in section 5.1 and active sender estimation
in section 5.2. The RTCP reporting interval for audio stream and video
stream should be calculated according to the rules in RFC3550 section
6.2 respectively. When RTCP reception reports for audio stream and
video stream are sent in different intervals, in order to decrease the
number of RTCP packet to be sent, it is more desirable to transmit
reception report for audio stream and reception report for video
stream in the same compound RTCP packet and set RTCP reporting
interval for audio stream as a multiple of RTCP reporting intervals
for video stream.</t>
<t>For example, if the RTCP Reporting Interval for audio stream is
more than one RTCP Reporting Interval for video but less than two RTCP
Reporting Intervals for video, it is RECOMMENDED the RTCP Reporting
Interval for audio stream be chosen as one RTCP Reporting Interval for
video.</t>
<t>If the RTCP Reporting Interval for audio stream is more than two
RTCP Reporting Intervals for video but less than three RTCP Reporting
Intervals for video, it is RECOMMENDED the RTCP Reporting Interval for
audio stream be chosen as two RTCP Reporting Intervals for video.</t>
<t>If the RTCP Reporting Interval for audio stream is more than three
RTCP Reporting Intervals for video but less than four RTCP Reporting
Intervals for video, it is RECOMMENDED the RTCP Reporting Interval for
audio stream be chosen as three RTCP Reporting Intervals for
video.</t>
</section>
<section title="RTCP reception report">
<t>Multiple-Source Endpoint SHOULD NOT send reception reports from one
of its source about all the other local sources of its own. RTP
application SHOULD provide a means to identify multiple-source
endpoint as in fact being sources from the same RTP node.</t>
<t>Multiple-Source Endpoint SHOULD combine RTCP reception reports into
a single compound RTCP packet without exceeding the maximum
transmission unit (MTU) of the network path.</t>
</section>
</section>
<section title="Security Considerations">
<t>TBC.</t>
</section>
<section title="IANA Considerations">
<t>This document has no actions for IANA.</t>
</section>
</middle>
<back>
<references title="Normative References">
<reference anchor="RFC2119">
<front>
<title abbrev="RFC Key Words">Key words for use in RFCs to Indicate
Requirement Levels</title>
<author fullname="Scott Bradner" initials="S." surname="Bradner">
<organization>Harvard University</organization>
<address>
<postal>
<street>1350 Mass. Ave.</street>
<street>Cambridge</street>
<street>MA 02138</street>
</postal>
<phone>- +1 617 495 3864</phone>
<email>sob@harvard.edu</email>
</address>
</author>
<date month="March" year="1997" />
<area>General</area>
<keyword>keyword</keyword>
<abstract>
<t>In many standards track documents several words are used to
signify the requirements in the specification. These words are
often capitalized. This document defines these words as they
should be interpreted in IETF documents. Authors who follow these
guidelines should incorporate this phrase near the beginning of
their document: <list>
<t>The key words "MUST", "MUST NOT", "REQUIRED", "SHALL",
"SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described
in RFC 2119.</t>
</list></t>
<t>Note that the force of these words is modified by the
requirement level of the document in which they are used.</t>
</abstract>
</front>
</reference>
<reference anchor="RFC3550">
<front>
<title>RTP: A Transport Protocol for Real-Time Applications</title>
<author fullname="Henning Schulzrinne" initials="H."
surname="Schulzrinne">
<organization>Columbia University</organization>
</author>
<date month="July" year="2003" />
</front>
<seriesInfo name="RFC" value="3550" />
<format type="TXT" />
</reference>
<reference anchor="RFC3556">
<front>
<title>Session Description Protocol (SDP) Bandwidth Modifiers for
RTP Control Protocol (RTCP) Bandwidth</title>
<author fullname="S.Casner" initials="S." surname="Casner">
<organization></organization>
</author>
<date month="July" year="2003" />
</front>
<seriesInfo name="RFC" value="3556" />
<format type="TXT" />
</reference>
<reference anchor="BUNDLE">
<front>
<title>Extended RTP Profile for Real-time Transport Control Protocol
(RTCP)-Based Feedback (RTP/AVPF)</title>
<author fullname="C.Holmberg" initials="C." surname="Holmberg">
<organization></organization>
</author>
<author fullname="H. Alvestrand" initials="H." surname="Alvestrand">
<organization></organization>
</author>
<date month="August" year="2012" />
</front>
<seriesInfo name="ID"
value="draft-ietf-mmusic-sdp-bundle-negotiation-01" />
<format type="TXT" />
</reference>
</references>
<references title="Informative References">
<reference anchor="I-D.westerlund-avtcore-multiplex-architecture">
<front>
<title>Guidelines for using the Multiplexing Features of RTP</title>
<author fullname="M.Westerlund" initials="M." surname="Westerlund">
<organization></organization>
</author>
<date month="July" year="2012" />
</front>
<seriesInfo name="ID"
value="draft-westerlund-avtcore-multiplex-architecture-02" />
<format type="TXT" />
</reference>
</references>
</back>
</rfc>
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