Network Working Group M. Petit-Huguenin
Internet-Draft Unaffiliated
Updates: 3550 (if approved) G. Zorn, Ed.
Intended status: Standards Track Network Zen
Expires: November 11, 2012 May 10, 2012
Support for Multiple Clock Rates in an RTP Session
draft-ietf-avtext-multiple-clock-rates-05
Abstract
This document clarifies the RTP specification when different clock
rates are used in an RTP session. It also provides guidance on how
to interoperate with legacy RTP implementations that use multiple
clock rates.
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 . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Legacy RTP . . . . . . . . . . . . . . . . . . . . . . . . . . 4
3.1. Different SSRC . . . . . . . . . . . . . . . . . . . . . . 4
3.2. Same SSRC . . . . . . . . . . . . . . . . . . . . . . . . 5
3.2.1. Monotonic timestamps . . . . . . . . . . . . . . . . . 5
3.2.2. Non-monotonic timestamps . . . . . . . . . . . . . . . 6
4. Recommendations . . . . . . . . . . . . . . . . . . . . . . . 7
4.1. RTP Sender (with RTCP) . . . . . . . . . . . . . . . . . . 7
4.2. RTP Sender (without RTCP) . . . . . . . . . . . . . . . . 7
4.3. RTP Receiver . . . . . . . . . . . . . . . . . . . . . . . 8
5. Security Considerations . . . . . . . . . . . . . . . . . . . 8
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 8
8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 9
8.1. Normative References . . . . . . . . . . . . . . . . . . . 9
8.2. Informative References . . . . . . . . . . . . . . . . . . 9
Appendix A. Using a Fixed Clock Rate . . . . . . . . . . . . . . 10
Appendix B. Behavior of Legacy Implementations . . . . . . . . . 10
B.1. libccrtp 2.0.2 . . . . . . . . . . . . . . . . . . . . . . 10
B.2. libmediastreamer0 2.6.0 . . . . . . . . . . . . . . . . . 10
B.3. libpjmedia 1.0 . . . . . . . . . . . . . . . . . . . . . . 10
B.4. Android RTP stack 4.0.3 . . . . . . . . . . . . . . . . . 10
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 10
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1. Introduction
The clock rate is a parameter of the payload format. It is often
defined as been the same as the sampling rate but it is not always
the case (see e.g. the G722 and MPA audio codecs [RFC3551]).
An RTP sender can switch between different payloads during the
lifetime of an RTP session and because clock rates are defined by
payload types, it is possible that the clock rate also varies during
an RTP session. Schulzrinne, et al. [RFC3550] lists using multiple
clock rates as one of the reasons to not use different payloads on
the same SSRC but unfortunately this advice was not always followed
and some RTP implementations change the payload in the same SSRC even
if the different payloads use different clock rates.
This creates three problems:
o The method used to calculate the RTP timestamp field in an RTP
packet is underspecified.
o When the same SSRC is used for different clock rates, it is
difficult to know what clock rate was used for the RTP timestamp
field in an RTCP SR packet.
o When the same SSRC is used for different clock rates, it is
difficult to know what clock rate was used for the interarrival
jitter field in an RTCP RR packet.
Table 1 contains a non-exhaustive list of fields in RTCP packets that
uses a clock rate as unit:
+---------------------+------------------+-----------+
| Field name | RTCP packet type | Reference |
+---------------------+------------------+-----------+
| RTP timestamp | SR | [RFC3550] |
| Interarrival jitter | RR | [RFC3550] |
| min_jitter | XR Summary Block | [RFC3611] |
| max_jitter | XR Summary Block | [RFC3611] |
| mean_jitter | XR Summary Block | [RFC3611] |
| dev_jitter | XR Summary Block | [RFC3611] |
| Interarrival jitter | IJ | [RFC5450] |
| RTP timestamp | SMPTETC | [RFC5484] |
| Jitter | RSI Jitter Block | [RFC5760] |
| Median jitter | RSI Stats Block | [RFC5760] |
+---------------------+------------------+-----------+
Table 1
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This document first tries to list in Section 3 and subsections all of
the algorithms known to be used in existing RTP implementations at
the time of writing. These sections are not normative.
Section 4 and subsections then recommend a unique algorithm that
modifies RFC 3550. These sections are normative.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
In addition, this document uses the following terms:
Clock rate The multiplier used to convert from a wallclock value
in seconds to an equivalent RTP timestamp value
(without the fixed random offset). Note that RFC 3550
uses various terms like "clock frequency", "media
clock rate", "timestamp unit", "timestamp frequency",
and "RTP timestamp clock rate" as synonymous to clock
rate.
RTP Sender A logical network element that sends RTP packets,
sends RTCP SR packets, and receives RTCP RR packets.
RTP Receiver A logical network element that receives RTP packets,
receives RTCP SR packets, and sends RTCP RR packets.
3. Legacy RTP
The following sections describe the various ways legacy RTP
implementations behave when multiple clock rates are used. Legacy
RTP refers to RFC 3550 without the modifications introduced by this
document.
3.1. Different SSRC
One way of managing multiple clock rates is to use a different SSRC
for each different clock rate, as in this case there is no ambiguity
on the clock rate used by fields in the RTCP packets. This method
also seems to be the original intent of RTP as can be deduced from
points 2 and 3 of section 5.2 of RFC 3550.
On the other hand changing the SSRC can be a problem for some
implementations designed to work only with unicast IP addresses,
where having multiple SSRCs is considered a corner case. Lip
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synchronization can also be a problem in the interval between the
beginning of the new stream and the first RTCP SR packet. This is
not different than what happen at the beginning of the RTP session
but it can be more annoying for the end-user.
3.2. Same SSRC
The simplest way of managing multiple clock rates is to use the same
SSRC for all the payload types regardless of the clock rates.
Unfortunately there is no clear definition on how the RTP timestamp
should be calculated in this case. The following subsections present
the algorithms used in the field.
3.2.1. Monotonic timestamps
This method of calculating the RTP timestamp ensures that the value
increases monotonically. The formula used by this method is as
follows:
timestamp = previous_timestamp
+ (current_capture_time - previous_capture_time)
* current_clock_rate
The problem with this method is that the jitter calculation on the
receiving side gives an invalid result during the transition between
two clock rates, as shown in Table 2. The capture and arrival time
are in seconds, starting at the beginning of the capture of the first
packet; clock rate is in Hz; the RTP timestamp does not include the
random offset; the transit, jitter, and average jitter use the clock
rate as unit.
+-------+-------+-----------+---------+---------+--------+----------+
| Capt. | Clock | RTP | Arrival | Transit | Jitter | Average |
| time | rate | timestamp | time | | | jitter |
+-------+-------+-----------+---------+---------+--------+----------+
| 0 | 8000 | 0 | 0.1 | 800 | | |
| 0.02 | 8000 | 160 | 0.12 | 800 | 0 | 0 |
| 0.04 | 8000 | 320 | 0.14 | 800 | 0 | 0 |
| 0.06 | 8000 | 480 | 0.16 | 800 | 0 | 0 |
| 0.08 | 16000 | 800 | 0.18 | 2080 | 480 | 30 |
| 0.1 | 16000 | 1120 | 0.2 | 2080 | 0 | 28 |
| 0.12 | 16000 | 1440 | 0.22 | 2080 | 0 | 26 |
| 0.14 | 8000 | 1600 | 0.24 | 320 | 720 | 70 |
| 0.16 | 8000 | 1760 | 0.26 | 320 | 0 | 65 |
+-------+-------+-----------+---------+---------+--------+----------+
Table 2
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Calculating the correct transit time on the receiving side can be
done by using the following formulas:
1. current_time_capture = current_timestamp - previous_timestamp) /
current_clock_rate + previous_time_capture
2. transit = current_clock_rate * (time_arrival -
current_time_capture)
3. previous_time_capture = current_time_capture
The main problem with this method, in addition to the fact that the
jitter calculation described in RFC 3550 cannot be used, is that is
it dependent on the previous RTP packets, packets that can be
reordered or lost in the network.
3.2.2. Non-monotonic timestamps
An alternate way of generating the RTP timestamps is to use the
following formula:
timestamp = capture_time * clock_rate
With this formula, the jitter calculation is correct but the RTP
timestamp values are no longer increasing monotonically as shown in
Table 3. RFC 3550 states that "[t]he sampling instant MUST be
derived from a clock that increments monotonically[...]" but nowhere
says that the RTP timestamp must increment monotonically.
+-------+-------+-----------+---------+---------+--------+----------+
| Capt. | Clock | RTP | Arrival | Transit | Jitter | Average |
| time | rate | timestamp | time | | | jitter |
+-------+-------+-----------+---------+---------+--------+----------+
| 0 | 8000 | 0 | 0.1 | 800 | | |
| 0.02 | 8000 | 160 | 0.12 | 800 | 0 | 0 |
| 0.04 | 8000 | 320 | 0.14 | 800 | 0 | 0 |
| 0.06 | 8000 | 480 | 0.16 | 800 | 0 | 0 |
| 0.08 | 16000 | 1280 | 0.18 | 1600 | 0 | 0 |
| 0.1 | 16000 | 1600 | 0.2 | 1600 | 0 | 0 |
| 0.12 | 16000 | 1920 | 0.22 | 1600 | 0 | 0 |
| 0.14 | 8000 | 1120 | 0.24 | 800 | 0 | 0 |
| 0.16 | 8000 | 1280 | 0.26 | 800 | 0 | 0 |
+-------+-------+-----------+---------+---------+--------+----------+
Table 3
The advantage with this method is that it works with the jitter
calculation described in RFC 3550, as long as the correct clock rates
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are used. It seems that this is what most implementations are using.
4. Recommendations
The following subsections describe behavioral recommendations for RTP
senders (with and without RTCP) and RTP recievers
4.1. RTP Sender (with RTCP)
An RTP Sender with RTCP turned on MUST use a different SSRC for each
different clock rate. An RTCP BYE MUST be sent and a new SSRC MUST
be used if the clock rate switches back to a value already seen in
the RTP stream.
To accelerate lip synchronization, the next compound RTCP packet sent
by the RTP sender MUST contain multiple SR packets, the first one
containing the mapping for the current clock rate and the next SR
packets containing the mapping for the other clock rates seen during
the last period.
The RTP extension defined in Perkins & Schierl [RFC6051] MAY be used
to accelerate the synchronization.
4.2. RTP Sender (without RTCP)
An RTP Sender with RTCP turned off (i.e. by setting the RS and RR
bandwidth modifiers [RFC3556] to 0) SHOULD use a different SSRC for
each different clock rate but MAY use different clock rates on the
same SSRC as long as the RTP timestamp without the random offset is
calculated as explained below:
Each time the clock rate changes, the start_offset and capture_start
values are calculated with the following formulas:
start_offset = (capture_time - capture_start)
* previous_clock_rate
capture_start = capture_time
For the first RTP packet, the values are initialized with the
following values:
start_offset = 0
capture_start = capture_time
After eventually updating these values, the RTP timestamp is
calculated with the following formula:
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timestamp = (capture_time - capture_start) * clock_rate
+ start_offset
Note that in all the formulas, capture_time is the first instant the
new timestamp rate is used.
4.3. RTP Receiver
An RTP Receiver MUST calculate the jitter using the following
formula:
D(i,j) = (arrival_time_j * clock_rate_i - timestamp_j)
- (arrival_time_i * clock_rate_i - timestamp_i)
An RTP Receiver MUST be able to handle a compound RTCP packet with
multiple SR packets.
For interoperability with legacy RTP implementations, an RTP receiver
MAY use the information in two consecutive SR packets to calculate
the clock rate used, i.e. if Ni is the NTP timestamp for the SR
packet i, Ri the RTP timestamp for the SR packet i and Nj and Rj the
NTP timestamp and RTP timestamp for the previous SR packet j, then
the clock rate can be guessed as the closest to (Ri - Rj) / (Ni -
Nj).
5. Security Considerations
This document is not believed to effect the security of the RTP
sessions described here in any way.
6. IANA Considerations
This document requires no IANA actions.
7. Acknowledgements
Thanks to Colin Perkins, Ali C. Begen, Harald Alvestrand and Magnus
Westerlund for their comments, suggestions and questions that helped
to improve this document.
Thanks to Robert Sparks and the attendees of SIPit 26 for the survey
on multiple clock rates interoperability.
This document was written with the xml2rfc tool described in Rose
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[RFC2629].
8. References
8.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
8.2. Informative References
[I-D.ietf-avt-variable-rate-audio]
Wenger, S. and C. Perkins, "RTP Timestamp Frequency for
Variable Rate Audio Codecs",
draft-ietf-avt-variable-rate-audio-00 (work in progress),
October 2004.
[RFC2629] Rose, M., "Writing I-Ds and RFCs using XML", RFC 2629,
June 1999.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
[RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth",
RFC 3556, July 2003.
[RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control
Protocol Extended Reports (RTCP XR)", RFC 3611,
November 2003.
[RFC5450] Singer, D. and H. Desineni, "Transmission Time Offsets in
RTP Streams", RFC 5450, March 2009.
[RFC5484] Singer, D., "Associating Time-Codes with RTP Streams",
RFC 5484, March 2009.
[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.
[RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
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Flows", RFC 6051, November 2010.
Appendix A. Using a Fixed Clock Rate
An alternate way of fixing the multiple clock rates issue was
proposed in [I-D.ietf-avt-variable-rate-audio]. This document
proposed to define a unified clock rate, but the proposal was
rejected at IETF 61.
Appendix B. Behavior of Legacy Implementations
B.1. libccrtp 2.0.2
This library uses the formula described in Section 3.2.2.
Note that this library uses gettimeofday(2) which is not guaranteed
to increment monotonically, like when the clock is adjusted by NTP.
B.2. libmediastreamer0 2.6.0
This library (which uses the oRTP library) uses the formula described
in Section 3.2.2.
Note that in some environments this library uses gettimeofday(2)
which is not guaranteed to increment monotonically.
B.3. libpjmedia 1.0
This library uses the formula described in Section 3.2.2.
B.4. Android RTP stack 4.0.3
This library changes the SSRC each time the format changes, as
described in Section 3.1.
Authors' Addresses
Marc Petit-Huguenin
Unaffiliated
Email: petithug@acm.org
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Glen Zorn (editor)
Network Zen
227/358 Thanon Sanphawut
Bang Na, Bangkok 10260
Thailand
Phone: +66 (0) 87-0404617
Email: glenzorn@gmail.com
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