Internet Engineering Task Force (IETF) X. Duan

Internet Engineering Task Force (IETF) X. Duan
Request for Comments: 5993 S. Wang
Category: Standards Track China Mobile Communications Corporation
ISSN: 2070-1721 M. Westerlund
K. Hellwig
I. Johansson
Ericsson AB
October 2010

RTP Payload Format for
Global System for Mobile Communications Half Rate (GSM-HR)

Abstract

This document specifies the payload format for packetization of
Global System for Mobile Communications Half Rate (GSM-HR) speech
codec data into the Real-time Transport Protocol (RTP). The payload
format supports transmission of multiple frames per payload and
packet loss robustness methods using redundancy.

Status of This Memo

This is an Internet Standards Track document.

This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 5741.

Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/rfc5993.

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RFC 5993 RTP Payload Format for GSM-HR October 2010

Copyright Notice

Copyright (c) 2010 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
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.

Table of Contents

1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Conventions Used in This Document . . . . . . . . . . . . . . 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 . . . . . . . . . . . . . . 8
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 . . . . . . . . . . . . . . . . . 9
6. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
6.1. 3 Frames . . . . . . . . . . . . . . . . . . . . . . . . . 10
6.2. 3 Frames with Lost Frame in the Middle . . . . . . . . . . 11
7. Payload Format Parameters . . . . . . . . . . . . . . . . . . 11
7.1. Media Type Definition . . . . . . . . . . . . . . . . . . 12
7.2. Mapping to SDP . . . . . . . . . . . . . . . . . . . . . . 13
7.2.1. Offer/Answer Considerations . . . . . . . . . . . . . 14
7.2.2. Declarative SDP Considerations . . . . . . . . . . . . 14
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 15
9. Congestion Control . . . . . . . . . . . . . . . . . . . . . . 15
10. Security Considerations . . . . . . . . . . . . . . . . . . . 15
11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 16
12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 16
12.1. Normative References . . . . . . . . . . . . . . . . . . . 16
12.2. Informative References . . . . . . . . . . . . . . . . . . 17

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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 payload format capabilities. The payload format is
specified in Section 5. Examples can be found in Section 6. The
media type specification and its mappings to SDP, and considerations
when using the Session Description Protocol (SDP) offer/answer
procedures are then specified. The document ends with considerations
related to 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 provided back in
1999 to 2000 in early draft versions of what was later published as
RFC 3551. RFC 3551 was based on a later version of the Audio-Visual
Profile (AVP) draft, which did not provide 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 Used in This Document

This document uses the normal IETF bit-order representation. Bit
fields in figures are read left to right and then down. The leftmost
bit in each field is the most significant. The numbering starts from
0 and ascends, where bit 0 will be the most significant.

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].

3. GSM Half Rate

The Global System for Mobile Communications (GSM) network provides
with mobile communication services for nearly 3 billion users
(statistics as of 2008). The GSM Half Rate (GSM-HR) codec is one of
the speech codecs 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.

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The GSM-HR codec has a frame length of 20 ms, with narrowband speech
sampled at 8000 Hz, 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 talkspurt and about
every 480 ms during speech pauses to allow for a decent comfort noise
(CN) quality on the 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 the uplink from the mobile station about
every 480 ms. In the downlink to the mobile station, when they are
generated by the encoder of the GSM radio subsystem, SID frames are
sent every 20 ms to the GSM base station, which then picks only one
every 480 ms for downlink radio transmission. For other
applications, like transport over IP, it is more appropriate to send
the SID frames less often than every 20 ms, 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 160 ms.

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 to indicate unsent 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 RFC 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.

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--+--------+--------+--------+--------+--------+--------+--------+--
| 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, signaling has been defined to allow the
sender to voluntarily bound the buffering and delay requirements. If
nothing is signaled, the use of this mechanism is allowed and
unbounded. For a certain timestamp, the receiver may acquire
multiple copies of a frame containing encoded audio data. The cost
of this scheme is bandwidth, and the receiver delay is 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 regarding 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]. The payload
format described in this document uses the header fields in a manner
consistent with that specification.

The duration of one speech frame is 20 ms. The sampling frequency is
8000 Hz, 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

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20-ms frame 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 a speech frame in one packet and as a
SID frame or as a 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 mandate 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

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Each ToC element is one octet and corresponds to one speech frame;
the number of ToC elements is thus equal to the number of speech
frames (including SID frames and No_Data frames). Each ToC entry
represents a consecutive speech or SID or No_Data frame. The
timestamp value for ToC element (and corresponding speech frame data)
N within the payload is (RTP timestamp field + (N-1)*160) mod 2^32.
The format of the ToC element 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 elements follow; 0 denotes
the last ToC element.

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
the case of Speech frames, with the unused 79 bits all set 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.

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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 their order of occurrence. The first
bit (b1) of the first parameter is placed in the most significant bit
(MSB) (bit 0) 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 least significant bit (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 all of the remaining 79 bits to "1".

5.3. Implementation Considerations

An application implementing this payload format MUST understand all
the payload parameters that are 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 160 ms, i.e., every 8th frame, during silent
periods. Other SID transmission intervals may occur due to gateways
to other systems that use 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.

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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.

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6. Examples

A few examples below highlight the payload format.

6.1. 3 Frames

Below is a basic example of the aggregation of 3 consecutive speech
frames into a single packet.

The first 24 bits are ToC elements.

Bit 0 is '1', as another ToC element follows.
Bits 1..3 are 000 = Good speech frame
Bits 4..7 are 0000 = Reserved
Bit 8 is '1', as another ToC element follows.
Bits 9..11 are 000 = Good speech frame
Bits 12..15 are 0000 = Reserved
Bit 16 is '0'; no more ToC elements follow.
Bits 17..19 are 000 = Good speech frame
Bits 20..23 are 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|
+-+-+-+-+-+-+-+-+

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6.2. 3 Frames with Lost Frame in the Middle

Below is an example of a 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 elements.

Bit 0 is '1', as another ToC element follows.
Bits 1..3 are 000 = Good speech frame
Bits 4..7 are 0000 = Reserved
Bit 8 is '1', as another ToC element follows.
Bits 9..11 are 111 = No_Data frame
Bits 12..15 are 0000 = Reserved
Bit 16 is '0'; no more ToC elements follow.
Bits 17..19 are 000 = Good speech frame
Bits 20..23 are 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 uses
[RFC4288] as a template. Note: Media subtype names are case-
insensitive.

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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-known speech codec. This media type
registration covers real-time transfer via RTP.

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 integers 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 of RFC 4288
[RFC4288].

Security considerations:

See Section 10 of RFC 5993.

Interoperability considerations:

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.

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Published specifications:

RFC 5993, 3GPP TS 46.002

Applications that use this media type:

Real-time audio applications like voice over IP and
teleconference.

Additional information: none

Person & email address to contact for further information:

Ingemar Johansson <[email protected]>

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.

Authors:

Xiaodong Duan <[email protected]>

Shuaiyu Wang <[email protected]>

Magnus Westerlund <[email protected]>

Ingemar Johansson <[email protected]>

Karl Hellwig <[email protected]>

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.

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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.

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 the Real Time Streaming Protocol
(RTSP) [RFC2326] or the Session Announcement Protocol (SAP)
[RFC2974], the parameters SHALL be interpreted as follows:

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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 than what has been declared. More than
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 it has been
registered 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 profiles, e.g.,
"RTP/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 security considerations discussed in the RTP
specification [RFC3550], and in any applicable RTP profile. The main
security considerations for the RTP packet carrying the RTP payload
format defined within this memo are confidentiality, integrity, and
source authenticity. Confidentiality is achieved by encryption of
the RTP payload, and integrity of the RTP packets through a suitable
cryptographic integrity protection mechanism. A cryptographic system
may also allow the authentication of the source of the payload. A
suitable security mechanism for this RTP payload format should
provide confidentiality, integrity protection, and at least source
authentication capable of determining whether or not an RTP packet is
from a member of the RTP session.

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Note that the appropriate mechanism to provide security to RTP and
payloads following this may vary. It is dependent on the
application, the transport, and the signaling protocol employed.
Therefore, a single mechanism is not sufficient, although if
suitable, the usage of the Secure Real-time Transport Protocol (SRTP)
[RFC3711] is recommended. Other mechanisms that may be used are
IPsec [RFC4301] and Transport Layer Security (TLS) [RFC5246] (e.g.,
for RTP over TCP), but other alternatives may also exist.

This RTP payload format and its media 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; nor
does the RTP payload format contain any active content.

11. Acknowledgements

The authors 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 for useful input and comments.

12. References

12.1. 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.

Duan, et al. Standards Track [Page 16]

RFC 5993 RTP Payload Format for GSM-HR October 2010

[TS46.002] 3GPP, "Half rate speech; Half rate speech processing
functions", 3GPP TS 46.002, June 2007, <http://
www.3gpp.org/ftp/Specs/archive/46_series/46.002/
46002-700.zip>.

[TS46.020] 3GPP, "Half rate speech; Half rate speech transcoding",
3GPP TS 46.020, June 2007, <http://www.3gpp.org/ftp/
Specs/archive/46_series/46.020/46020-700.zip>.

12.2. 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.

[RFC4301] Kent, S. and K. Seo, "Security Architecture for the
Internet Protocol", RFC 4301, December 2005.

[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.

[RFC5246] Dierks, T. and E. Rescorla, "The Transport Layer Security
(TLS) Protocol Version 1.2", RFC 5246, August 2008.

Duan, et al. Standards Track [Page 17]

RFC 5993 RTP Payload Format for GSM-HR October 2010

Authors' Addresses

Xiaodong Duan
China Mobile Communications Corporation
53A, Xibianmennei Ave., Xuanwu District
Beijing, 100053
P.R. China
EMail: [email protected]

Shuaiyu Wang
China Mobile Communications Corporation
53A, Xibianmennei Ave., Xuanwu District
Beijing, 100053
P.R. China
EMail: [email protected]

Magnus Westerlund
Ericsson AB
Farogatan 6
Stockholm, SE-164 80
Sweden
Phone: +46 8 719 0000
EMail: [email protected]

Karl Hellwig
Ericsson AB
Ericsson Allee 1
52134 Herzogenrath
Germany
Phone: +49 2407 575-2054
EMail: [email protected]

Ingemar Johansson
Ericsson AB
Laboratoriegrand 11
SE-971 28 Lulea
Sweden
Phone: +46 73 0783289
EMail: [email protected]

Duan, et al. Standards Track [Page 18]