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RTP Payload Format for AC-3 Audio

The information below is for an old version of the document that is already published as an RFC.
Document Type
This is an older version of an Internet-Draft that was ultimately published as RFC 4184.
Authors Jason Flaks , Todd Hager , Brian Link
Last updated 2013-03-02 (Latest revision 2005-06-28)
Replaces draft-flaks-avt-rtp-ac3
RFC stream Internet Engineering Task Force (IETF)
Intended RFC status Proposed Standard
Additional resources Mailing list discussion
Stream WG state (None)
Document shepherd (None)
IESG IESG state Became RFC 4184 (Proposed Standard)
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Consensus boilerplate Unknown
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Responsible AD Allison J. Mankin
Send notices to,
Audio/Video Transport                                           B. Link
Internet-Draft                                                 T. Hager
Expires: December 2005                               Dolby Laboratories
                                                               J. Flaks
                                                  Microsoft Corporation
                                                              June 2005
                  RTP Payload Format for AC-3 Audio

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This document describes an RTP payload format for transporting AC-3 
encoded audio data.  AC-3 is a high quality, multichannel audio coding 
system used in US HDTV, DVD, cable and satellite television and other 
media.  The RTP payload format as presented in this document includes 
support for data fragmentation.

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1. Introduction 

AC-3 is a high quality audio codec designed to encode multiple channels
of audio into a low bit-rate format.  AC-3 achieves its large 
compression ratios via encoding a multiplicity of channels as a single 
entity.  Dolby Digital, which is a branded version of AC-3, encodes up 
to 5.1 channels of audio.
AC-3 has been adopted as an audio compression scheme for many consumer 
and professional applications.  It is a mandatory audio codec for 
DVD-video, Advanced Television Standards Committee (ATSC) digital 
terrestrial television and Digital Living Network Alliance (DLNA) home 
networking, as well as an optional multichannel audio format for 

There is a need to stream AC-3 data over IP networks.  RTP provides a 
mechanism for stream synchronization and hence serves as the best 
transport solution for AC-3, which is a codec primarily used in 
audio-for-video applications.  Applications for streaming AC-3 include 
streaming movies from a home media server to a display, video on 
demand, and multichannel Internet radio.    

Section 2 gives a brief overview of the AC-3 algorithm.  Section 3 
specifies values for fields in the RTP header, while Section 4 
specifies the AC-3 payload format, itself.  Section 5 discusses MIME 
types and SDP usage.  Security considerations are covered in Section 6,
Congestion Control in Section 7, and IANA considerations in Section 8.
References are given in Sections 9 and 10.  

The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 
document are to be interpreted as described in RFC 2119 [RFC2119]. 

2. Overview of AC-3

AC-3 can deliver up to 5.1 channels of audio at data rates 
approximately equal to half of one PCM channel [ATSC], [1994AC3], 
[1996AC3].  The ".1" refers to a band-limited, optional, low-frequency 
enhancement channel.  AC-3 was designed for signals sampled at rates 
of 32, 44.1, or 48 kHz.  Data rates can vary between 32 kbps and 
640 kbps, depending the number of channels and desired quality.

AC-3 exploits psychoacoustic phenomena that cause a significant 
fraction of the information contained in a typical audio signal to be 
inaudible.  Substantial data reduction occurs via the removal of 
inaudible information contained in an audio stream.  Source coding 
techniques are further used to reduce the data rate.

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Like most perceptual coders, AC-3 operates in the frequency domain.  A 
512-point TDAC transform is taken with 50% overlap, providing 256 new 
frequency samples.  Frequency samples are then converted to exponents 
and mantissas.  Exponents are differentially encoded.  Mantissas are 
allocated a varying number of bits depending on the audibility of the 
spectral components associated with them.  Audibility is determined 
via a masking curve.  Bits for mantissas are allocated from a global 
bit pool.

2.1 AC-3 Bit Stream 

AC-3 bit streams are organized into synchronization frames.  Each AC-3 
frame contains a Sync Information (SI) field, a Bit Stream 
Information (BSI) field, and 6 audio blocks (AB), each representing 
256 PCM samples for each channel.  The frame ends with an optional 
auxiliary data field (AUX) and an error correction field (CRC).  The 
entire frame represents the time duration of 1536 PCM samples across 
all coded channels [ATSC].  AC-3 encodes audio sampled at 32 kHz, 
44.1 kHz, and 48 kHz.  From Annex A, Part 2, of [ATSC], the time 
duration of an AC-3 frame varies with the sampling rate as follows:

      Sampling rate          Frame duration
         48   kHz                32    ms
         44.1 kHz        approx. 34.83 ms
         32   kHz                48    ms

Figure 1 shows the AC-3 frame format. 

|SI |BSI|  AB0  |  AB1  |  AB2  |  AB3  |  AB4  |  AB5  |AUX|CRC|

                    Figure 1. AC-3 Frame Format

The Synchronization Information field contains information needed to 
acquire and maintain codec synchronization.  The Bit Stream 
Information field contains parameters that describe the coded audio 
service [ATSC].  Each audio block contains fields that 
indicate the use of various coding tools: block switching, dither, 
coupling, and exponent strategy.  They also contain metadata, 
optionally used to enhance the playback, such as dynamic range control.
Figure 2 shows the structure of an AC-3 audio block.  Note that field 
sizes vary depending on the coded data.  

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|  Block  |Dither |Dynamic    |Coupling |Coupling     |Exponent |
|  Switch |Flags  |Range Ctrl |Strategy |Coordinates  |Strategy |
|     Exponents       | Bit Allocation  |        Mantissas      |
|                     | Parameters      |                       |

                  Figure 2. AC-3 Audio Block Format

3. RTP Header Fields 

Payload Type (PT): The assignment of an RTP payload type for this 
packet format is outside the scope of this document; it is specified 
by the RTP profile under which this payload format is used, or 
signaled dynamically out-of-band (e.g., using SDP).

Marker (M) bit: The M bit is set to one to indicate that the RTP 
packet payload contains at least one complete AC-3 frame or contains 
the final fragment of an AC-3 frame. 

Extension (X) bit: Defined by the RTP profile used. 

Timestamp: A 32-bit word that corresponds to the sampling instant for 
the first AC-3 frame in the RTP packet.  Packets containing fragments 
of the same frame MUST have the same time stamp.  The timestamp of the 
first RTP packet sent SHOULD be selected at random; thereafter it 
increases linearly according to the number of samples included in each 
frame (i.e. by 1536 for each frame).

4. RTP AC-3 Payload Format 

This payload format is defined for AC-3, as defined in the main part 
and Annex D of [ATSC]. It is not defined for E-AC-3, as defined in 
Annex E of [ATSC] and MUST not be used to carry E-AC-3.

According to [RFC2736], RTP payload formats should contain an integral 
number of application data units (ADUs).  An ADU shall be equivalent 
to an AC-3 frame.  Each RTP payload MUST start with the two-byte 
payload header followed by an integral number of complete AC-3 frames, 
or a single fragment of an AC-3 frame. 

If an AC-3 frame exceeds the MTU for a network, it SHOULD be 
fragmented for transmission within an RTP packet.  Section 4.2 
provides guidelines for creating frame fragments.

4.1 Payload-Specific Header 

There is a two-octet Payload Header at the beginning of each payload.  

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4.1.1 Payload Header 

Each AC-3 RTP payload MUST begin with the following payload header.  
Figure 3 shows the format of this header.

                  0                   1             
                  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 
                 |    MBZ    | FT|       NF      | 

                  Figure 3. AC-3 RTP Payload Header

Must Be Zero (MBZ): Bits marked MBZ SHALL be set to the value zero and 
SHALL be ignored by receivers. The bits are reserved for future 

Frame Type (FT): This two-bit field indicates the type of frame(s) 
present in the payload. It takes the following values:
          0 - One or more complete frames.
          1 - Initial fragment of frame which includes the first 
              5/8ths of the frame.  (See Section 4.2.)
          2 - Initial fragment of frame, which does not include the 
              first 5/8ths of the frame.
          3 - Fragment of frame other than initial fragment.  (Note 
              that M bit in RTP header is set for final fragment.)

Number of frames/fragments(NF): An 8-bit field whose meaning depends 
on the Frame Type (FT) in this payload. For complete frames (FT of 0), 
it is used to indicate the number of AC-3 frames in the RTP payload.  
For frame fragments (FT of 1, 2, or 3), it is used to indicate the 
number fragments (and therefore packets) that make up the current 
frame.  NF MUST be identical for packets containing fragments of the 
same frame.

Figure 4 shows the full AC-3 RTP payload format. 

      +-+-+-+-+-+-+-+-+-+-+-+-+-+- .. +-+-+-+-+-+-+-+
      | Payload | Frame | Frame |     | Frame |
      | Header  |  (1)  |  (2)  |     |  (n)  |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+- .. +-+-+-+-+-+-+-+

                   Figure 4. Full AC-3 RTP payload

When receiving AC-3 payloads with FT = 0 and more than a single frame 
(NF > 0), a receiver needs to use the "frmsizecod" field in the 
synchronization information (syncinfo) block in each AC-3 frame to 
determine the frame's length.  That way a receiver can determine 
the boundary of the next frame.  Note that the frame length varies 
from frame to frame in some circumstances.

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4.2 Fragmentation of AC-3 Frames 

The size of an AC-3 frame depends on the sample rate of the audio and 
the data rate used by the encoder (which are indicated in the 
"Synchronization Information" header in the AC-3 frame.)  The size of 
a frame, for a given sample rate and data rate, is specified in 
Table 5.18 ("Frame Size Code Table") of [ATSC].  This table shows 
that AC-3 frames range in size from a minimum of 128 bytes to a 
maximum of 3840 bytes.  If the size of an AC-3 frame exceeds the MTU 
size, the frame SHOULD be fragmented at the RTP level.  The 
fragmentation MAY be performed at any byte boundary in the frame. RTP 
packets containing fragments of the same AC-3 frame SHALL be sent in 
consecutive order, from first to last fragment.  This enables a 
receiver to assemble the fragments in correct order.

When an AC-3 frame is fragmented, it MAY be fragmented such that at 
least the first 5/8ths of the frame data is in the first fragment.  
This provides greater resilience to packet loss.  This initial 
portion of any frame is guaranteed to contain the data necessary to 
decode the first two blocks of the frame.  Any frame fragments other 
than those containing the first 5/8ths of frame data are only 
decodable once the complete frame is received.  The 5/8ths point of 
the frame is defined in Table 7.34 ("5/8_frame Size Table") of 

5. Types and Names

5.1 Media Type Registration

This registration uses the template defined in [DRAFT-FREED] and 
follows RFC 3555 [RFC3555].

Type name:                         audio

Subtype name:                      ac3

Required parameters:
        rate: The RTP timestamp clock rate which is equal to the audio 
        sampling rate.  Permitted rates are 32000, 44100, and 48000.

Optional parameters:
        channels: From a sender, the maximum number of channels present
        in the AC3 stream.  From a receiver, the maximum number of 
        output channels the receiver will deliver.  This MUST be a 
        number between 1 and 6. The LFE (".1") channel MUST be counted 
        as one channel.  Note that the channel order used in AC-3 
        differs from the channel order scheme in [RFC3551].  The AC-3 
        channel order scheme can be found in Table 5.8 of [ATSC].

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        ptime: See RFC 2327 [RFC2327].  

        maxptime: See RFC 3267 [RFC3267].

Encoding considerations:
        This media type is framed (see section 4.8 in [DRAFT-FREED]) 
        and contains binary data.

Security considerations:
        See Section 6 of this document. 

Interoperability considerations:

Published specification:
        This payload format specification and see [ATSC].

Applications which use this media type: 
        Multichannel audio compression of audio and audio for video.

Additional Information:
Magic number(s):
        The first two octets of an AC-3 frame are always the 
        synchronization word, which has the hex value 0x0B77. 

Person & email address to contact for further information:
        Brian Link <>
        IETF AVT working group.

Intended Usage:

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/Change controller:  
        IETF Audio/Video Transport Working Group delegated from 
        the IESG.

5.2 SDP Usage 

The information carried in the MIME media type specification has a 
specific mapping to fields in the Session Description Protocol (SDP) 
[RFC2327], which is commonly used to describe RTP sessions.  When SDP 
is used to specify sessions employing AC-3, the mapping is as follows:

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   o   The Media type  ("audio") goes in SDP "m=" as the media name.

   o   The Media subtype  ("ac3") goes in SDP "a=rtpmap" 
       as the encoding name.

   o   The required parameter "rate" also goes in "a=rtpmap" as the 
       clock rate, optionally followed by the parameter "channel".

   o   The optional parameters "ptime" and "maxptime" go in the SDP
       "a=ptime" and "a=maxptime" attributes, respectively.

An example of the SDP data for AC-3:
   m=audio 49111 RTP/AVP 100
   a=rtpmap:100 ac3/48000/6

Certain considerations are needed when SDP is used to perform 
offer/answer exchanges [RFC3264].  The "rate" is a symmetric parameter 
and the answer MUST use the same value or remove the payload type.

The "channels" parameter is declarative and indicates, for reconly or 
sendrecv, the desired channel configuration to receive, and for 
sendonly, the intended channel configuration to transmit.  All 
receivers are capable of receiving any of the defined channel 
configurations and the parameter exchange might be used to help 
optimize the transmission to the number of channels the receiver 
requests.  If the "channels" parameter is omitted, a default maximum 
value of 6 is implied.  "ptime" and "maxptime" are negotiated as 
defined for "ptime" in RFC 3264 [RFC3264].

6. Security Considerations 

The payload format described in this document is subject to the 
security considerations defined in RTP [RFC3550] and in any applicable 
RTP profile (e.g. [RFC3551]).  To protect the user's privacy and any 
copyrighted material, confidentiality protection would have to be 
applied.  To also protect against modification by intermediate 
entities and ensure the authenticity of the stream, integrity 
protection and authentication would be required.  Confidentiality, 
integrity protection, and authentication have to be solved by a 
mechanism external to this payload format, e.g., SRTP [RFC3711].

The AC-3 format is designed so that the validity of data frames can 
determined by decoders.  The required decoder response to a malformed 
frame is to discard the malformed data and conceal the errors in the 
audio output until a valid frame is detected and decoded.  This is 
expected to prevent crashes and other abnormal decoder behavior in 
response to errors or attacks.

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7. Congestion Control

The general congestion control considerations for transporting RTP 
data apply to AC-3 audio over RTP as well, see RTP [RFC3550], and any 
applicable RTP profile (e.g., [RFC3551]).  

AC-3 encoders may use a range of bit rates to encode audio data, so 
it is possible to adapt network bandwidth by adjusting the encoder 
bit rate in real time or by having multiple copies of content encoded 
at different bit rates.  Additionally, packing more frames in each RTP 
payload can reduce the number of packets sent and hence the overhead 
from IP/UDP/RTP headers, at the expense of increased delay and reduced 
error robustness against packet losses.

8. IANA Considerations

Registration of a new media subtype for AC-3 is requested (see 
Section 5.)

9. Normative References 

[RFC2119] Bradner, S., "Key Words for use in RFCs to Indicate 
Requirement Levels", RFC 2119, Internet Engineering Task Force, 
March 1997. 

[ATSC] U.S. Advanced Television Systems Committee (ATSC), "ATSC 
Standard: Digital Audio Compression (AC-3), Revision B," Doc A/52B, 
June 2005. 

[RFC2327] Handley, M. and Jacobson, V., "SDP: Session Description 
Protocol," RFC 2327, Internet Engineering Task Force, April 1998 

[RFC3550] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobsen, 
"RTP: A Transport Protocol for Real-Time Applications", RFC 3550, 
STD 64, July 2003.

[RFC3264] Rosenberg, J. and Schulzrinne, H., "An Offer/Answer Model 
with the Session Description Protocol (SDP)", RFC 3264, Internet 
Engineering Task Force, June 2002.

[RFC3267] Sjoberg, J., et. al., "Real-Time Transport Protocol (RTP) 
Payload Format and File Storage Format for the Adaptive Multi-Rate 
(AMR) and Adaptive Multi-Rate Wideband (AMR-WB) Audio Codecs", 
RFC 3267, Internet Engineering Task Force, June 2002.

[RFC3555] Casner, S. and Hoschka, P., "MIME Type Registration of RTP 
Payload Formats", RFC 3555, Internet Engineering Task Force, July 2003.

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10. Informative References

[RFC2736] Handley, M. and Perkins, C., "Guidelines for Writers of RTP 
Payload Format Specifications," RFC 2736, Internet Engineering Task 
Force, December 1999.  

[RFC3551] Schulzrinne, H., Casner, S., "RTP Profile for Audio and 
Video Conferences with Minimal Control", RFC 3551, Internet 
Engineering Task Force, July 2003.

[1994AC3] Todd, C. et. al, "AC-3: Flexible Perceptual Coding for Audio 
Transmission and Storage," Preprint 3796, Presented at the 96th 
Convention of the Audio Engineering Society, May 1994. 

[1996AC3] Fielder, L. et. al, "AC-2 and AC-3: Low-Complexity 
Transform-Based Audio Coding," Collected Papers on Digital Audio 
Bit-Rate Reduction, pp. 54-72, Audio Engineering Society, 
September 1996.  

[RFC3711] Baugher, M. et. al, "The Secure Real-time Transport Protocol 
(SRTP)", RFC 3711, March 2004.
[DRAFT-FREED] Freed, N. and Klensin, J., "Media Type Specifications 
and Registration Procedures", draft-freed-media-type-reg-04, 
April 2005.

Authors' Addresses

Brian Link
Dolby Laboratories 
100 Potrero Ave 
San Francisco, CA 94103 

Phone: +1 415 558 0200

Todd Hager 
Dolby Laboratories 
100 Potrero Ave 
San Francisco, CA 94103 

Phone: +1 415 558 0136

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Jason Flaks
Microsoft Corporation
1 Microsoft Way
Redmond, WA 98052

Phone: +1 425 722 2543

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