Audio/Video Transport Working Group Q. Wu
Internet-Draft R. Even
Intended status: Standards Track Huawei
Expires: April 25, 2013 October 22, 2012
Bandwidth and RTCP timing issues for multi-source endpoint
draft-wu-avtcore-multiplex-multisource-endpoint-01.txt
Abstract
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.
Status of this Memo
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
2.1. Standards Language . . . . . . . . . . . . . . . . . . . . 4
3. Use Cases for multi-source host communications . . . . . . . . 5
3.1. One to one Communication between multi-source endpoints . 5
3.2. Multi-party Communication between multi-source
endpoints . . . . . . . . . . . . . . . . . . . . . . . . 6
3.3. Communication between multi-source endpoint
multiplexing each medium in separate RTP session and
multiple-source endpoint multiplexing all mediums in
one RTP session . . . . . . . . . . . . . . . . . . . . . 7
4. Discuss . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
5. Recommendations . . . . . . . . . . . . . . . . . . . . . . . 10
5.1. Choosing RTCP Bandwidth . . . . . . . . . . . . . . . . . 10
5.2. Maintaining the number of session members and senders . . 11
5.3. RTCP Reporting Interval calculation . . . . . . . . . . . 12
5.4. RTCP reception report . . . . . . . . . . . . . . . . . . 13
6. Security Considerations . . . . . . . . . . . . . . . . . . . 14
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 15
8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 16
8.1. Normative References . . . . . . . . . . . . . . . . . . . 16
8.2. Informative References . . . . . . . . . . . . . . . . . . 16
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 17
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1. Introduction
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.
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.
This document provides recommendations for multi-source host sending
multiple media types in the same session.
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2. Terminology
2.1. Standards Language
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
Multi-Source Endpoint
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.
Single-Source Endpoint
End system with single source generated from one host described in
RFC3550.
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3. Use Cases for multi-source host communications
3.1. One to one Communication between multi-source endpoints
Multiple-Source Multiple-Source
Endpoint1 Endpoint2
+--------+ +--------+
| | Common Transport | |
| | | |
| |Transport flow1(v1,a1) |
SSRC-V1---| |------------------>| |--- SSRC-V2
| | | |
| | | |
| | | |
| | | |
SSRC-A1---| |Transport flow1(v2,a2) |
| |<------------------| +----SSRC-A2
| | | |
| | | |
| | | |
+--------+ +--------+
Figure 1
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.
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.
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
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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.
3.2. Multi-party Communication between multi-source endpoints
+---+
| |
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
| | +-----------+
| |
| |
+---+
Figure 2
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.
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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.
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.
3.3. Communication between multi-source endpoint multiplexing each
medium in separate RTP session and multiple-source endpoint
multiplexing all mediums in one RTP session
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
| | | |
| | | |
| | | |
+--------+ +--------+
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
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transport(i.e.,transport flow1)to multiple-source endpoint2.
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4. Discuss
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.
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.
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5. Recommendations
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.
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.
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
1. 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.
2. 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.
3. 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.
Multi-source endpoints multiplexing each media type in separate RTP
session SHOULD be treated as multiple endpoints for each media type.
Editor's Note: It will be useful to discuss why we recommend to have
a specific algorithm and see how it works with mixers and translators
in the future version.
5.1. Choosing RTCP Bandwidth
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
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sent over the same transport and the media data rate of audio stream
is far less than video stream.
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%.
RTCP bandwidth for video = session bandwidth *5%
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.
RTCP bandwidth for audio stream SHOULD be calculated based on the
following formula:
RTCP bandwidth for audio = ((audio codec maximum bitrate*20%)+ audio
codec maximum bitrate)*5%
Here a 20% RTP packet overhead is added to the data rate to calculate
the required RTCP bandwidth.
5.2. Maintaining the number of session members and senders
As described in section 6.2.1 of RFC3550, Calculation of the RTCP
packet interval mainly depends upon the number of members. For an
endpoint with multiple sources multiplexing all media types in the
same RTP session, it is more desirable to set the number of sender
for such endpoint as one and the number of receiver as one. However
according to RFC3550, each source within such endpoint is considered
valid until multiple packets carrying the new SSRC have been
received, or until an SDES RTCP packet containing a CNAME for that
SSRC has been received.
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.
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
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which is sending reception reports.
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.
5.3. RTCP Reporting Interval calculation
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.
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.
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.
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.
Editor's Note: The problem is not as simple as just varying the RTCP
bandwidth fraction for the audio sources in an RTP session. Varying
the reporting interval in this way for some participants can lead to
timeouts. You'd also need to revise the timeout rules,
reconsideration rules, etc.
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5.4. RTCP reception report
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.
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.
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6. Security Considerations
TBC.
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7. IANA Considerations
This document has no actions for IANA.
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8. References
8.1. Normative References
[BUNDLE] Holmberg, C. and H. Alvestrand, "Extended RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/AVPF)",
ID draft-ietf-mmusic-sdp-bundle-negotiation-01,
August 2012.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", March 1997.
[RFC3550] Schulzrinne, H., "RTP: A Transport Protocol for Real-Time
Applications", RFC 3550, July 2003.
[RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth",
RFC 3556, July 2003.
8.2. Informative References
[I-D.ietf-avtcore-multi-media-rtp-session-00]
Westerlund, M., "Multiple Media Types in an RTP Session",
ID draft-ietf-avtcore-multi-media-rtp-session-00,
October 2012.
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Authors' Addresses
Qin Wu
Huawei
101 Software Avenue, Yuhua District
Nanjing, Jiangsu 210012
China
Email: sunseawq@huawei.com
Roni Even
Huawei
14 David Hamelech
Tel, Aviv 64953
Israel
Email: ron.even.tlv@gmail.com
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