Network Working Group X. Duan
Internet-Draft S. Wang
Intended status: Standards Track China Mobile Communications
Expires: October 17, 2009 Corporation
M. Westerlund
K. Hellwig
I. Johansson
Ericsson AB
Apr 15, 2009
RTP Payload format for GSM-HR
draft-ietf-avt-rtp-gsm-hr-00
Status of this Memo
This Internet-Draft is submitted to IETF in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF), its areas, and its working groups. Note that
other groups may also distribute working documents as Internet-
Drafts.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
The list of current Internet-Drafts can be accessed at
http://www.ietf.org/ietf/1id-abstracts.txt.
The list of Internet-Draft Shadow Directories can be accessed at
http://www.ietf.org/shadow.html.
This Internet-Draft will expire on October 17, 2009.
Copyright Notice
Copyright (c) 2009 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents in effect on the date of
publication of this document (http://trustee.ietf.org/license-info).
Please review these documents carefully, as they describe your rights
and restrictions with respect to this document.
Duan, et al. Expires October 17, 2009 [Page 1]
Internet-Draft RTP Payload format for GSM-HR Apr 2009
Abstract
This document specifies the RTP payload format for packetization of
the GSM Half-Rate speech codec.
Requirements 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].
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Conventions . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. GSM Half Rate . . . . . . . . . . . . . . . . . . . . . . . . 3
4. Payload format Capabilities . . . . . . . . . . . . . . . . . 4
4.1. Use of Forward Error Correction (FEC) . . . . . . . . . . 4
5. Payload format . . . . . . . . . . . . . . . . . . . . . . . . 5
5.1. RTP Header Usage . . . . . . . . . . . . . . . . . . . . . 6
5.2. Payload Structure . . . . . . . . . . . . . . . . . . . . 6
5.2.1. Encoding of Speech Frames . . . . . . . . . . . . . . 7
5.2.2. Encoding of Silence Description Frames . . . . . . . . 8
5.3. Implementation Considerations . . . . . . . . . . . . . . 8
5.3.1. Transmission of SID frames . . . . . . . . . . . . . . 8
5.3.2. Receiving Redundant Frames . . . . . . . . . . . . . . 8
5.3.3. Decoding Validation . . . . . . . . . . . . . . . . . 8
6. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
6.1. 3 frames . . . . . . . . . . . . . . . . . . . . . . . . . 9
6.2. 3 Frames with lost frame in the middle . . . . . . . . . . 10
7. Payload Format Parameters . . . . . . . . . . . . . . . . . . 10
7.1. Media Type Definition . . . . . . . . . . . . . . . . . . 11
7.2. Mapping to SDP . . . . . . . . . . . . . . . . . . . . . . 12
7.2.1. Offer/Answer Considerations . . . . . . . . . . . . . 13
7.2.2. Declarative SDP Considerations . . . . . . . . . . . . 13
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 14
9. Congestion Control . . . . . . . . . . . . . . . . . . . . . . 14
10. Security Considerations . . . . . . . . . . . . . . . . . . . 14
10.1. Confidentiality . . . . . . . . . . . . . . . . . . . . . 15
10.2. Authentication and Integrity . . . . . . . . . . . . . . . 15
11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 15
12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 15
12.1. Informative References . . . . . . . . . . . . . . . . . . 15
12.2. Normative References . . . . . . . . . . . . . . . . . . . 16
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 16
Duan, et al. Expires October 17, 2009 [Page 2]
Internet-Draft RTP Payload format for GSM-HR Apr 2009
1. Introduction
This document specifies the payload format for packetization of GSM
Half Rate (GSM-HR) codec [TS46.002] encoded speech signals into the
Real-time Transport Protocol (RTP) [RFC3550]. The payload format
supports transmission of multiple frames per payload and packet loss
robustness methods using redundancy.
This document starts with conventions, a brief description of the
codec, and the payload formats capabilities. The payload format is
specified in Section 5. Examples can be found in Section 6. The
media type and its mappings to SDP, usage in SDP offer/answer is then
specified. The document ends with considerations around congestion
control and security.
This document registers a media type (audio/gsm-hr-08) for the Real-
time Transport protocol (RTP) payload format for the GSM-HR codec.
Note: This format is not compatible with the one that was drafted
back in 1999 to 2000 in the Internet drafts:
draft-ietf-avt-profile-new-05 to draft-ietf-avt-profile-new-09. A
later version of profile draft was published as RFC 3551 without any
specification of the GSM-HR payload format. To avoid a possible
conflict with this older format, the media type of the payload format
specified in this document has a media type name that is different
from (audio/gsm-hr).
2. Conventions
This document uses the normal IETF bit-order representation. Bit
fields in figures are read left to right and then down. The left
most bits in each field is the most significant. The numbering
starts on 0 and ascends, where bit 0 will be the most significant.
3. GSM Half Rate
The Global System for Mobile Communication (GSM) network provides
mobile communication services for nearly 3 billion users (status
2008). The GSM Half Rate Codec (GSM-HR) is one of the speech codecs
that are used in GSM networks. GSM-HR denotes the Half-Rate speech
codec as specified in [TS46.002].
Note: for historical reasons these 46-series specifications are
internally referenced as 06-series. A simple mapping applies, for
example 46.020 is referenced as 06.20 and so on.
The GSM-HR codec has a frame length of 20 ms, with narrowband speech
Duan, et al. Expires October 17, 2009 [Page 3]
Internet-Draft RTP Payload format for GSM-HR Apr 2009
sampled at 8 kHz, i.e. 160 samples per frame. Each speech frame is
compressed into 112 bits of speech parameters, which is equivalent to
a bit rate of 5.6 kbit/s. Speech pauses are detected by a
standardized Voice Activity Detection (VAD). During speech pauses
the transmission of speech frames is inhibited. Silence Descriptor
(SID) frames are transmitted at the end of a talk spurt and about
every 480ms during speech pauses to allow for a decent Comfort Noise
(CN) quality at receiver side.
The SID frame generation in the GSM radio network is determined by
the GSM mobile station and the GSM radio subsystem. SID frames come
during speech pauses in uplink from the mobile station about every
480ms. In downlink to the mobile station, when they are generated by
the encoder of the GSM radio subsystem, SID frames are sent every
20ms to the GSM base station, which then picks only one every 480ms
for downlink radio transmission. For other applications, like
transport over IP, it is more appropriate to send the SID frames less
often than every 20ms, but 480 ms may be too sparse. We recommend as
a compromise that a GSM-HR encoder outside of the GSM radio network
(i.e. not in the GSM mobile station and not in the GSM radio
subsystem, but for example in the media gateway of the core network)
should generate and send SID frames every 160ms.
4. Payload format Capabilities
This RTP payload format carries one or more GSM-HR encoded frames,
either full voice or silence descriptor (SID), representing a mono
speech signal. To maintain synchronization or express not sent or
lost frames it has the capability to indicate No_Data frames.
4.1. Use of Forward Error Correction (FEC)
Generic forward error correction within RTP is defined, for example,
in RFC 5109 [RFC5109]. Audio redundancy coding is defined in RFC
2198 [RFC2198]. Either scheme can be used to add redundant
information to the RTP packet stream and make it more resilient to
packet losses, at the expense of a higher bit rate. Please see
either RFCs for a discussion of the implications of the higher bit
rate to network congestion.
In addition to these media-unaware mechanisms, this memo specifies an
optional to use GSM-HR specific form of audio redundancy coding,
which may be beneficial in terms of packetization overhead.
Conceptually, previously transmitted transport frames are aggregated
together with new ones. A sliding window can be used to group the
frames to be sent in each payload. Figure 1 below shows an example.
Duan, et al. Expires October 17, 2009 [Page 4]
Internet-Draft RTP Payload format for GSM-HR Apr 2009
--+--------+--------+--------+--------+--------+--------+--------+--
| f(n-2) | f(n-1) | f(n) | f(n+1) | f(n+2) | f(n+3) | f(n+4) |
--+--------+--------+--------+--------+--------+--------+--------+--
<---- p(n-1) ---->
<----- p(n) ----->
<---- p(n+1) ---->
<---- p(n+2) ---->
<---- p(n+3) ---->
<---- p(n+4) ---->
Figure 1: An example of redundant transmission
Here, each frame is retransmitted once in the following RTP payload
packet. f(n-2)...f(n+4) denote a sequence of audio frames, and p(n-
1)...p(n+4) a sequence of payload packets.
The mechanism described does not really require signaling at the
session setup. However, signalling has been defined to allow for the
sender to voluntarily bounding the buffering and delay requirements.
If nothing is signalled the use of this mechanism is allowed and
unbounded. For a certain timestamp, the receiver may receive
multiple copies of a frame containing encoded audio data. The cost
of this scheme is bandwidth and the receiver delay necessary to allow
the redundant copy to arrive.
This redundancy scheme provides a functionality similar to the one
described in RFC 2198, but it works only if both original frames and
redundant representations are GSM-HR frames. When the use of other
media coding schemes is desirable, one has to resort to RFC 2198.
The sender is responsible for selecting an appropriate amount of
redundancy based on feedback about the channel conditions, e.g., in
the RTP Control Protocol (RTCP) [RFC3550] receiver reports. The
sender is also responsible for avoiding congestion, which may be
exacerbated by redundancy (see Section 9 for more details).
5. Payload format
The format of the RTP header is specified in [RFC3550]. This payload
format uses the fields of the header in a manner consistent with that
specification.
The duration of one speech frame is 20 ms. The sampling frequency is
8kHz, corresponding to 160 speech samples per frame. An RTP packet
may contain multiple frames of encoded speech or SID parameters.
Each packet covers a period of one or more contiguous 20 ms frame
Duan, et al. Expires October 17, 2009 [Page 5]
Internet-Draft RTP Payload format for GSM-HR Apr 2009
intervals. During silence periods no speech packets are sent,
however SID packets are transmitted every now and then.
To allow for error resiliency through redundant transmission, the
periods covered by multiple packets MAY overlap in time. A receiver
MUST be prepared to receive any speech frame multiple times. A given
frame MUST NOT be encoded as speech frame in one packet and as SID
frame or as No_Data frame in another packet. Furthermore, a given
frame MUST NOT be encoded with different voicing modes in different
packets.
The rules regarding maximum payload size given in Section 3.2 of
[RFC5405] SHOULD be followed.
5.1. RTP Header Usage
The RTP timestamp corresponds to the sampling instant of the first
sample encoded for the first frame in the packet. The timestamp
clock frequency SHALL be 8000 Hz. The timestamp is also used to
recover the correct decoding order of the frames.
The RTP header marker bit (M) SHALL be set to 1 whenever the first
frame carried in the packet is the first frame in a talkspurt (see
definition of the talkspurt in section 4.1 of [RFC3551]). For all
other packets the marker bit SHALL be set to zero (M=0).
The assignment of an RTP payload type for the format defined in this
memo is outside the scope of this document. The RTP profiles in use
currently mandates binding the payload type dynamically for this
payload format.
The remaining RTP header fields are used as specified in RFC 3550
[RFC3550].
5.2. Payload Structure
The complete payload consists of a payload table of contents (ToC)
section, followed by speech data representing one or more speech
frames, SID frames or No_Data frames. The following diagram shows
the general payload format layout:
+-------------+-------------------------
| ToC section | speech data section ...
+-------------+-----------------------
Figure 2: General payload format layout
Each ToC is one octet and corresponds to one speech frame, the number
of ToC's is thus equal to the number of speech frames (including SID
Duan, et al. Expires October 17, 2009 [Page 6]
Internet-Draft RTP Payload format for GSM-HR Apr 2009
frames and No_Data frames). Each ToC entry represents a consecutive
speech or SID or No_Data frame. The timestamp value for ToC entry
(and corresponding speech frame data) N within the payload is (RTP
timestamp field + (N-1)*160) mod 2^32 . The format of the ToC octet
is as follows.
0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
|F| FT |R R R R|
+-+-+-+-+-+-+-+-+
Figure 3: The TOC element
F: Follow flag, 1 denotes that more ToC's follow, 0 denotes the last
ToC.
R: Reserved bits, MUST be set to zero and MUST be ignored by
receiver.
FT: Frame type
000 = Good Speech frame
001 = Reserved
010 = Good SID frame
011 = Reserved
100 = Reserved
101 = Reserved
110 = Reserved
111 = No_Data frame
The length of the payload data depends on the frame type:
Good Speech frame: The 112 speech data bits are put in 14 octets.
Good SID frame: The 33 SID data bits are put in 14 octets, as in
case of Speech frames, with the unused 79 bits set all to "1".
No data frame: Length of payload data is zero octets.
Frames marked in the GSM radio subsystem as "Bad Speech frame", "Bad
SID frame" or "No_Data frame" are not sent in RTP packets in order to
save bandwidth. They are marked as "No_Data frame", if they occur
within an RTP packet that carries more than one speech frame, SID
frame or No_Data frame.
5.2.1. Encoding of Speech Frames
The 112 bits of GSM-HR-coded speech (b1...b112) are defined in TS
46.020, Annex B [TS46.020], in the order of occurrence. The first
Duan, et al. Expires October 17, 2009 [Page 7]
Internet-Draft RTP Payload format for GSM-HR Apr 2009
bit (b1) of the first parameter is placed in bit 0 (the MSB) of the
first octet (octet 1) of the payload field; the second bit is placed
in bit 1 of the first octet and so on. The last bit (b112) is placed
in the LSB (bit 7) of octet 14.
5.2.2. Encoding of Silence Description Frames
The GSM-HR Codec applies a specific coding for silence periods in so
called SID frames. The coding of SID frames is based on the coding
of speech frames by using only the first 33 bits for SID parameters
and by setting the remaining 79 bits all to "1".
5.3. Implementation Considerations
An application implementing this payload format MUST understand all
the payload parameters that is defined in this specification. Any
mapping of the parameters to a signaling protocol MUST support all
parameters. So an implementation of this payload format in an
application using SDP is required to understand all the payload
parameters in their SDP-mapped form. This requirement ensures that
an implementation always can decide whether it is capable of
communicating when the communicating entities support this version of
the specification.
5.3.1. Transmission of SID frames
When using this RTP payload format the sender SHOULD generate and
send SID frames every 160ms, i.e. every 8th frame. Other SID
transmission intervals may occur due to gateways to other systems
that uses other transmission intervals.
5.3.2. Receiving Redundant Frames
The reception of redundant audio frames, i.e. more than one audio
frame from the same source for the same time slot, MUST be supported
by the implementation.
5.3.3. Decoding Validation
If the receiver finds a mismatch between the size of a received
payload and the size indicated by the ToC of the payload, the
receiver SHOULD discard the packet. This is recommended because
decoding a frame parsed from a payload based on erroneous ToC data
could severely degrade the audio quality.
Duan, et al. Expires October 17, 2009 [Page 8]
Internet-Draft RTP Payload format for GSM-HR Apr 2009
6. Examples
A few examples to highlight the payload format.
6.1. 3 frames
A basic example of the aggregation of 3 consecutive speech frames
into a single frame.
The first 24 bits are ToC fields.
Bit 0 is '1' as another ToC field follow.
Bits 1..3 is 000 = Good speech frame
Bits 4..7 is 0000 = Reserved
Bits 8 is '1' as another ToC field follow.
Bits 9..11 is 000 = Good speech frame
Bits 12..15 is 0000 = Reserved
Bit 16 is '0', no more ToC follows
Bits 17..19 is 000 = Good speech frame
Bits 20..23 is 0000 = Reserved
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1|0 0 0|0 0 0 0|1|0 0 0|0 0 0 0|0|0 0 0|0 0 0 0|b1 b8|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +
|b9 Frame 1 b40|
+ +
|b41 b72|
+ +
|b73 b104|
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|b105 b112|b1 b24|
+-+-+-+-+-+-+-+-+ +
|b25 Frame 2 b56|
+ +
|b57 b88|
+ +-+-+-+-+-+-+-+-+
|b89 b112|b1 b8|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +
|b9 Frame 3 b40|
+ +
|b41 b72|
+ +
|b73 b104|
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|b105 b112|
+-+-+-+-+-+-+-+-+
Duan, et al. Expires October 17, 2009 [Page 9]
Internet-Draft RTP Payload format for GSM-HR Apr 2009
6.2. 3 Frames with lost frame in the middle
An example of payload carrying 3 frames where the middle one is
No_Data, for example due to loss prior to transmission by the RTP
source.
The first 24 bits are ToC fields.
Bit 0 is '1' as another ToC field follow.
Bits 1..3 is 000 = Good speech frame
Bits 4..7 is 0000 = Reserved
Bits 8 is '1' as another ToC field follow.
Bits 9..11 is 111 = No_Data frame
Bits 12..15 is 0000 = Reserved
Bit 16 is '0', no more ToC follows
Bits 17..19 is 000 = Good speech frame
Bits 20..23 is 0000 = Reserved
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1|0 0 0|0 0 0 0|1|1 1 1|0 0 0 0|0|0 0 0|0 0 0 0|b1 b8|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +
|b9 Frame 1 b40|
+ +
|b41 b72|
+ +
|b73 b104|
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|b105 b112|b1 b24|
+-+-+-+-+-+-+-+-+ +
|b25 Frame 3 b56|
+ +
|b57 b88|
+ +-+-+-+-+-+-+-+-+
|b89 b112|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
7. Payload Format Parameters
This RTP payload format is identified using the media type "audio/
gsm-hr-08", which is registered in accordance with [RFC4855] and
using the template of [RFC4288]. Note: Media subtype names are case-
insensitive.
Duan, et al. Expires October 17, 2009 [Page 10]
Internet-Draft RTP Payload format for GSM-HR Apr 2009
7.1. Media Type Definition
The media type for the GSM-HR codec is allocated from the IETF tree
since GSM-HR is a well know speech codec. This media type
registration covers real-time transfer via RTP. The media subtype
name contains "-08" to avoid potential conflict with any earlier
drafts of GSM-HR RTP payload types that aren't bit compatible.
Note, reception of any unspecified parameter MUST be ignored by the
receiver to ensure that additional parameters can be added in the
future.
Type name: audio
Subtype name: GSM-HR-08
Required parameters: none
Optional parameters:
max-red: The maximum duration in milliseconds that elapses between
the primary (first) transmission of a frame and any redundant
transmission that the sender will use. This parameter allows a
receiver to have a bounded delay when redundancy is used. Allowed
values are between 0 (no redundancy will be used) and 65535. If
the parameter is omitted, no limitation on the use of redundancy
is present.
ptime: see [RFC4566].
maxptime: see [RFC4566].
Encoding considerations:
This media type is framed and binary, see section 4.8 in RFC4288
[RFC4288].
Security considerations:
See Section 10 of RFCXXXX.
Interoperability considerations:
Published specification:
RFC XXXX, 3GPP TS 46.002
Applications that use this media type:
Duan, et al. Expires October 17, 2009 [Page 11]
Internet-Draft RTP Payload format for GSM-HR Apr 2009
Real-time audio applications like voice over IP and
teleconference.
Additional information: none
Person & email address to contact for further information:
Ingemar Johansson <ingemar.s.johansson@ericsson.com>
Intended usage: COMMON
Restrictions on usage:
This media type depends on RTP framing, and hence is only defined
for transfer via RTP [RFC3550]. Transport within other framing
protocols is not defined at this time.
Author:
Xiaodong Duan <duanxiaodong@chinamobile.com>
Shuaiyu Wang <wangshuaiyu@chinamobile.com>
Magnus Westerlund <magnus.westerlund@ericsson.com>
Ingemar Johansson <ingemar.s.johansson@ericsson.com>
Karl Hellwig <karl.hellwig@ericsson.com>
Change controller:
IETF Audio/Video Transport working group delegated from the IESG.
7.2. Mapping to SDP
The information carried in the media type specification has a
specific mapping to fields in the Session Description Protocol (SDP)
[RFC4566], which is commonly used to describe RTP sessions. When SDP
is used to specify sessions employing the GSM-HR codec, the mapping
is as follows:
o The media type ("audio") goes in SDP "m=" as the media name.
o The media subtype (payload format name) goes in SDP "a=rtpmap" as
the encoding name. The RTP clock rate in "a=rtpmap" MUST be 8000,
and the encoding parameters (number of channels) MUST either be
explicitly set to 1 or omitted, implying a default value of 1.
Duan, et al. Expires October 17, 2009 [Page 12]
Internet-Draft RTP Payload format for GSM-HR Apr 2009
o The parameters "ptime" and "maxptime" go in the SDP "a=ptime" and
"a=maxptime" attributes, respectively.
o Any remaining parameters go in the SDP "a=fmtp" attribute by
copying them directly from the media type parameter string as a
semicolon-separated list of parameter=value pairs.
7.2.1. Offer/Answer Considerations
The following considerations apply when using SDP Offer-Answer
procedures to negotiate the use of GSM-HR payload in RTP:
o The SDP offerer and answerer MUST generate GSM-HR packets as
described by the offered parameters.
o In most cases, the parameters "maxptime" and "ptime" will not
affect interoperability; however, the setting of the parameters
can affect the performance of the application. The SDP offer-
answer handling of the "ptime" parameter is described in
[RFC3264]. The "maxptime" parameter MUST be handled in the same
way.
o The parameter "max-red" is a stream property parameter. For
sendonly or sendrecv unicast media streams, the parameter declares
the limitation on redundancy that the stream sender will use. For
recvonly streams, it indicates the desired value for the stream
sent to the receiver. The answerer MAY change the value, but is
RECOMMENDED to use the same limitation as the offer declares. In
the case of multicast, the offerer MAY declare a limitation; this
SHALL be answered using the same value. A media sender using this
payload format is RECOMMENDED to always include the "max-red"
parameter. This information is likely to simplify the media
stream handling in the receiver. This is especially true if no
redundancy will be used, in which case "max-red" is set to 0.
o Any unknown media type parameter in an offer SHALL be removed in
the answer.
7.2.2. Declarative SDP Considerations
In declarative usage, like SDP in RTSP [RFC2326] or SAP [RFC2974],
the parameters SHALL be interpreted as follows:
o The stream property parameter ("max-red") is declarative, and a
participant MUST follow what is declared for the session. In this
case it means that the receiver MUST be prepared to allocate
buffer memory for the given redundancy. Any transmissions MUST
NOT use more redundancy then what has been declared. More than
Duan, et al. Expires October 17, 2009 [Page 13]
Internet-Draft RTP Payload format for GSM-HR Apr 2009
one configuration may be provided if necessary by declaring
multiple RTP payload types; however, the number of types should be
kept small.
o Any "maxptime" and "ptime" values should be selected with care to
ensure that the session's participants can achieve reasonable
performance.
8. IANA Considerations
One media type (audio/gsm-hr-08) has been defined and needs
registration in the media types registry; see Section 7.1.
9. Congestion Control
The general congestion control considerations for transporting RTP
data apply; see RTP [RFC3550] and any applicable RTP profile like AVP
[RFC3551].
The number of frames encapsulated in each RTP payload highly
influences the overall bandwidth of the RTP stream due to header
overhead constraints. Packetizing more frames in each RTP payload
can reduce the number of packets sent and hence the header overhead,
at the expense of increased delay and reduced error robustness. If
forward error correction (FEC) is used, the amount of FEC-induced
redundancy needs to be regulated such that the use of FEC itself does
not cause a congestion problem.
10. Security Considerations
RTP packets using the payload format defined in this specification
are subject to the general security considerations discussed in RTP
[RFC3550] and any applicable profile such as AVP [RFC3551] or SAVP
[RFC3711]. As this format transports encoded audio, the main
security issues include confidentiality, integrity protection, and
data origin authentication of the audio itself. The payload format
itself does not have any built-in security mechanisms. Any suitable
external mechanisms, such as SRTP [RFC3711], MAY be used.
This payload format and the GSM-HR decoder do not exhibit any
significant non-uniformity in the receiver-side computational
complexity for packet processing, and thus are unlikely to pose a
denial-of-service threat due to the receipt of pathological data.
The payload format or the codec data does not contain any type of
active content such as scripts.
Duan, et al. Expires October 17, 2009 [Page 14]
Internet-Draft RTP Payload format for GSM-HR Apr 2009
10.1. Confidentiality
In order to ensure confidentiality of the encoded audio, all audio
data bits MUST be encrypted. There is less need to encrypt the
payload header or the table of contents since they only carry
information about the frame type. This information could also be
useful to a third party, for example, for quality monitoring.
10.2. Authentication and Integrity
To authenticate the sender of the audio-stream, an external mechanism
MUST be used. It is RECOMMENDED that such a mechanism protects both
the complete RTP header and the payload (audio and data bits). Data
tampering by a man-in-the-middle attacker could replace audio content
and also result in erroneous depacketization/decoding that could
lower the audio quality.
11. Acknowledgements
The author would like to thank Xiaodong Duan, Shuaiyu Wang, Rocky
Wang and Ying Zhang for their initial work in this area. Many thanks
also go to Tomas Frankkila, Karl Hellwig for useful input and
comments.
12. References
12.1. Informative References
[RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
September 1997.
[RFC2326] Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time
Streaming Protocol (RTSP)", RFC 2326, April 1998.
[RFC2974] Handley, M., Perkins, C., and E. Whelan, "Session
Announcement Protocol", RFC 2974, October 2000.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[RFC4288] Freed, N. and J. Klensin, "Media Type Specifications and
Registration Procedures", BCP 13, RFC 4288, December 2005.
Duan, et al. Expires October 17, 2009 [Page 15]
Internet-Draft RTP Payload format for GSM-HR Apr 2009
[RFC4855] Casner, S., "Media Type Registration of RTP Payload
Formats", RFC 4855, February 2007.
[RFC5109] Li, A., "RTP Payload Format for Generic Forward Error
Correction", RFC 5109, December 2007.
12.2. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
June 2002.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC5405] Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines
for Application Designers", BCP 145, RFC 5405,
November 2008.
[TS46.002]
3GPP, "Specification : 3GPP TS 46.002 http://www.3gpp.org/
ftp/Specs/archive/46_series/46.002/46002-700.zip",
June 2007.
[TS46.020]
3GPP, "Specification : 3GPP TS 46.020 http://www.3gpp.org/
ftp/Specs/archive/46_series/46.002/46020-700.zip",
June 2007.
Duan, et al. Expires October 17, 2009 [Page 16]
Internet-Draft RTP Payload format for GSM-HR Apr 2009
Authors' Addresses
Xiaodong Duan
China Mobile Communications Corporation
53A, Xibianmennei Ave., Xuanwu District
Beijing, 100053
P.R. China
Phone:
Fax:
Email: duanxiaodong@chinamobile.com
URI:
Shuaiyu Wang
China Mobile Communications Corporation
53A, Xibianmennei Ave., Xuanwu District
Beijing, 100053
P.R. China
Phone:
Fax:
Email: wangshuaiyu@chinamobile.com
URI:
Magnus Westerlund
Ericsson AB
Farogatan 6
Stockholm, SE-164 80
Sweden
Phone: +46 8 719 0000
Fax:
Email: magnus.westerlund@ericsson.com
URI:
Karl Hellwig
Ericsson AB
Kackertstrasse 7-9
52072 Aachen
Germany
Phone: +49 2407 575-2054
Email: karl.hellwig@ericsson.com
Duan, et al. Expires October 17, 2009 [Page 17]
Internet-Draft RTP Payload format for GSM-HR Apr 2009
Ingemar Johansson
Ericsson AB
Laboratoriegrand 11
SE-971 28 Lulea
SWEDEN
Phone: +46 73 0783289
Email: ingemar.s.johansson@ericsson.com
Duan, et al. Expires October 17, 2009 [Page 18]
