RFC 1890

RTP Profile for Audio and Video Conferences with Minimal Control

📅 published January 1996, from ietf

Okay bestie, let's get into it 💖🎀 This is RFC 1890, called RTP Profile for Audio and Video Conferences with Minimal Control 🦋. It was written by H. Schulzrinne of GMD Fokus, and it comes out of the Audio-Video Transport Working Group, which sits inside the Network Working Group 🌸. It's a Standards Track document dated January 1996. The title block doesn't list anything it obsoletes or updates, so it's standing on its own 💅.

Status of This Memo ✨

It's an Internet standards track protocol for the whole Internet community, and it's literally asking you to discuss it and send suggestions 🫶. For the standardization state and status of this protocol, check the current edition of the "Internet Official Protocol Standards" (STD 1) 📚. Distribution of this memo is unlimited, so share away 💖.

Abstract 🌈

Okay so this memo is a profile for using RTP (the real-time transport protocol, version 2) and its bestie RTCP in audio and video multiparticipant conferences with minimal control ☎️. It gives meaning to the generic RTP fields so they actually fit audio and video calls 🎥. The big highlight is that it defines a set of default mappings from payload type numbers to encodings 🔥.

It also describes how audio and video data may be carried within RTP 🎶. It defines a set of standard encodings and their names when used within RTP, but the encoding definitions don't depend on which transport mechanism you use 🙈. The descriptions point you to reference implementations and the detailed standards, and the whole thing is meant as a helper for implementors of audio, video and other real-time multimedia apps 🧚.

1. Introduction ✨

This profile fills in the parts of RTP that the RTP Version 2 protocol definition (RFC 1889) left unspecified 💅. It's meant for audio and video conferences with minimal session control, so there's no support for negotiating parameters or membership control 🫠. It's expected to be useful in sessions where no negotiation or membership control is used (for example, with the static payload types and the membership indications RTCP provides) 🌸. It may also be useful alongside a higher-level control protocol, though 🦋.

You use this profile just by using the right applications. There's no explicit indication of it by port number, protocol identifier or anything like that 🙃.

Other profiles MAY make different choices for the items specified here 🌈.

2. RTP and RTCP Packet Forms and Protocol Behavior 💅

The section "RTP Profiles and Payload Format Specification" lists a bunch of items that can be specified or modified in a profile, and this section covers those items 📝. Generally, this profile sticks with the default and/or recommended aspects of the RTP specification 💖.

  • RTP data header: The standard format of the fixed RTP data header is used (one marker bit) ✨.
  • Payload types: Static payload types are defined in Section 6 🎀.
  • RTP data header additions: No additional fixed fields are appended to the RTP data header 🙅‍♀️.
  • RTP data header extensions: No RTP header extensions are defined, but applications operating under this profile MAY use such extensions 🧩. So applications SHOULD NOT assume that the RTP header X bit is always zero, and they SHOULD be prepared to ignore the header extension 👀. If a header extension is defined in the future, that definition MUST specify the contents of the first 16 bits in such a way that multiple different extensions can be identified 🔐.
  • RTCP packet types: No additional RTCP packet types are defined by this profile specification 📦.
  • RTCP report interval: The suggested constants are to be used for the RTCP report interval calculation ⏰.
  • SR/RR extension: No extension section is defined for the RTCP SR or RR packet 🚫.
  • SDES use: Applications MAY use any of the SDES items described 🏷️. While CNAME information is sent every reporting interval, other items SHOULD be sent only every fifth reporting interval 🫶.
  • Security: The RTP default security services are also the default under this profile 🔒.
  • String-to-key mapping: A user-provided string ("pass phrase") gets hashed with the MD5 algorithm into a 16-octet digest 🔑. An n-bit key is pulled out of the digest by taking the first n bits from it. If several keys are needed with a total length of 128 bits or less (as for triple DES), they're extracted in order from that digest 🧁. The octet ordering is specified in RFC 1423, Section 2.2 📜. (Heads up: some DES implementations need the 56-bit key expanded into 8 octets by inserting an odd parity bit in the most significant bit of each octet, to go with each 7 bits of the key 🙈.)
  • Underlying protocol: The profile specifies the use of RTP over unicast and multicast UDP 🌐. (This doesn't stop these definitions from being used when RTP rides on other lower-layer protocols 🦋.)
  • Transport mapping: The standard mapping of RTP and RTCP to transport-level addresses is used 🗺️.
  • Encapsulation: No encapsulation of RTP packets is specified 🎁.
🚨 Pass phrase vibes: it's suggested that pass phrases are limited to ASCII letters, digits, the hyphen and white space, so there are fewer transcription errors when you share keys by phone, fax, telex or email 📞.

A pass phrase MAY be preceded by a specification of the encryption algorithm 🔐. Any characters up to the first slash (ASCII 0x2f) are taken as the name of the encryption algorithm. The encryption format specifiers SHOULD be drawn from RFC 1423 or from any additional identifiers registered with IANA ✅. If there's no slash, DES-CBC is assumed as the default 💅. The encryption algorithm specifier is case sensitive, so be careful with those capitals 🔠.

The pass phrase the user types gets transformed into a canonical form before the hash algorithm runs on it ✨. For that, we define return, tab, and vertical tab, plus all the characters in the Unicode space characters table 🌌. The transformation goes like this: (1) convert the input string to the ISO 10646 character set, using the UTF-8 encoding as specified in Annex P to ISO/IEC 10646-1:1993 (ASCII characters need no mapping, but ISO 8859-1 characters do) 🌍; (2) remove leading and trailing white space characters ✂️; (3) replace one or more contiguous white space characters with a single space (ASCII or UTF-8 0x20) 🫧; (4) convert all letters to lower case, and replace sequences of characters and non-spacing accents with a single character where possible 🍓. A minimum length of 16 key characters (after the transformation) SHOULD be enforced by the application, while applications MUST allow up to 256 characters of input 💜.

3. Registering Payload Types 🎀

This profile defines a set of standard encodings and their payload types for use within RTP 🌷. Any other encodings and payload types get registered with the Internet Assigned Numbers Authority (IANA) 🏛️. When you register a new encoding or payload type, you SHOULD provide the following information 📝:

  • name and description of the encoding, in particular the RTP timestamp clock rate; the names defined here are 3 or 4 characters long so they can be represented compactly if needed 🔤;
  • who has change control over the encoding (for example, ISO, CCITT/ITU, other international standardization bodies, a consortium, or a particular company or group of companies) 👑;
  • any operating parameters or profiles ⚙️;
  • a reference to a further description, if one exists, for example (in order of preference) an RFC, a published paper, a patent filing, a technical report, documented source code or a computer manual 📚;
  • for proprietary encodings, contact information (postal and email address) 💌;
  • the payload type value for this profile, if necessary (see below) 🧚.

Not every encoding used by RTP needs a static payload type 🦋. Non-RTP stuff outside the scope of this memo (like directory services or invitation protocols) MAY be used to set up a dynamic mapping between a payload type from the range 96-127 and an encoding 🪄. For implementor convenience, this profile also describes encodings that don't currently have a static payload type assigned 🫶.

The payload type space is pretty small, so new static payload types only get assigned if these conditions are met 🥺:

  • The encoding is of interest to the Internet community at large 🌍.
  • It offers benefits compared to existing encodings and/or is required for interoperation with existing, widely deployed conferencing or multimedia systems 🔥.
  • The description is sufficient to build a decoder 🛠️.

4. Audio 🎧

4.1. Encoding-Independent Recommendations 🎀

okay bestie, this part is all the stuff that stays the same no matter which codec you pick 💅 If your app goes silent and sends zero packets during quiet moments, the very first packet after a silence gap (the start of a talkspurt) gets its marker bit set in the RTP data header 🦋 Apps without silence suppression just leave that bit at zero ✨

The RTP clock rate that makes your timestamps doesn't care how many channels you have or what encoding you picked. It's simply the number of sampling periods per second ⏰ So for an N-channel encoding, each sampling period (like 1/8000 of a second) spits out N samples. ngl the naming is standard but kinda confusing, because the total samples per second is actually the sampling rate times the channel count 🙈

When you've got multiple audio channels, they get numbered left to right, starting at one. In RTP audio packets, the lower-numbered channels come first, then the higher ones 👑 For more than two channels, follow the AIFF-C audio interchange format convention [1], using this notation:

  • l left
  • r right
  • c center
  • S surround
  • F front
  • R rear

Here's the chart, iykyk 💖

channels    description                 channel
                            1     2     3     4     5     6
___________________________________________________________
2           stereo          l     r
3                           l     r     c
4           quadrophonic    Fl    Fr    Rl    Rr
4                           l     c     r     S
5                           Fl    Fr    Fc    Sl    Sr
6                           l     lc    c     r     rc    S

Samples for all channels belonging to a single sampling instant must be within the same packet. How the samples from different channels get interleaved depends on the encoding. General guidelines are in Section 4.2 and 4.3 🧚

The sampling frequency should be drawn from this set: 8000, 11025, 16000, 22050, 24000, 32000, 44100 and 48000 Hz. Fun fact, the Apple Macintosh computers have native sample rates of 22254.54 and 11127.27, and those can be converted to 22050 and 11025 with acceptable quality by dropping 4 or 2 samples in a 20 ms frame 🍎 Most audio encodings are only defined for a more restricted set of sampling frequencies though. Receivers should be prepared to accept multi-channel audio, but they may choose to only play a single channel 🎧

These are the default operating parameters. Applications should be prepared to handle other values. The ranges are there to give application writers guidance, so a set of applications following these guidelines can interoperate without any extra negotiation 🫶 They aren't meant to restrict anything for apps that can negotiate a set of interoperable parameters, like through a conference control protocol.

For packetized audio, the default packetization interval should be 20 ms long, unless the encoding description says otherwise. The packetization interval sets the minimum end-to-end delay. Longer packets mean less header overhead, but more delay, and packet loss gets more noticeable 😭 For non-interactive stuff like lectures or links with severe bandwidth constraints, a higher packetization delay may be totally appropriate 🌸 A receiver should accept packets representing between 0 and 200 ms of audio data, which keeps buffer sizing reasonable for the receiver 💜

4.2. Guidelines for Sample-Based Audio Encodings 🍓

In sample-based encodings, each audio sample is represented by a fixed number of bits. Inside the compressed audio data, the codes for individual samples may span octet boundaries. An RTP audio packet can hold any number of audio samples, as long as the bits per sample times the samples per packet adds up to a whole number of octets. Fractional encodings produce less than one octet per sample 🌈

The duration of an audio packet is determined by how many samples are in the packet ⏱️

For sample-based encodings producing one or more octets per sample, samples from different channels sampled at the same instant get packed into consecutive octets. For example, with a two-channel encoding, the octet sequence goes (left channel, first sample), (right channel, first sample), (left channel, second sample), (right channel, second sample), and so on. For multi-octet encodings, the octets are sent in network byte order, which means most significant octet first 🔥

The packing for sample-based encodings producing less than one octet per sample depends on the specific encoding 🤷‍♀️

4.3. Guidelines for Frame-Based Audio Encodings 🦋

Frame-based encodings take a fixed-length block of audio and turn it into another block of compressed data, which is usually also fixed length. With frame-based encodings, the sender can choose to bundle several of these frames into a single message. The receiver can tell how many frames are in a message, because the frame duration is defined as part of the encoding ✨

For frame-based codecs, the channel order is defined for the whole block. So for two-channel audio, the right and left samples are coded independently, and the encoded frame for the left channel comes before the one for the right channel 👑

All frame-oriented audio codecs should be able to encode and decode several consecutive frames within a single packet. Since the frame size for these codecs is given, there's no need for a separate name for the same encoding just because it uses a different number of frames per packet 💅

4.4. Audio Encodings 📻

Here's the cheat sheet of the standard audio encodings 🧁

        encoding    sample/frame    bits/sample    ms/frame
        ____________________________________________________
        1016        frame           N/A            30
        DVI4        sample          4
        G721        sample          4
        G722        sample          8
        G728        frame           N/A            2.5
        GSM         frame           N/A            20
        L8          sample          8
        L16         sample          16
        LPC         frame           N/A            20
        MPA         frame           N/A
        PCMA        sample          8
        PCMU        sample          8
        VDVI        sample          var.

              Table 1: Properties of Audio Encodings

The characteristics of the standard audio encodings are in Table 1, and their payload types are listed in Table 2 💖

4.4.1. 1016

Encoding 1016 is a frame-based encoding that uses code-excited linear prediction (CELP), and it's specified in Federal Standard FED-STD 1016 [2,3,4,5] 🎙️

The U. S. DoD's Federal-Standard-1016 based 4800 bps code excited linear prediction voice coder version 3.2 (CELP 3.2) Fortran and C simulation source codes are available for worldwide distribution at no charge. They come on DOS diskettes but are configured to compile on Sun SPARC stations. You can get them from Bob Fenichel, National Communications System, Washington, D.C. 20305, phone +1-703-692-2124, fax +1-703-746-4960 📠

4.4.2. DVI4

DVI4 is specified, with pseudo-code, in [6] as the IMA ADPCM wave type. There's also a spec called "DVI ADPCM Wave Type" in the Microsoft Developer Network Development Library CD ROM, which Microsoft publishes quarterly. You'll find the relevant section under Product Documentation, SDKs, Multimedia Standards Update, New Multimedia Data Types and Data Techniques, Revision 3.0, April 15, 1994. However, the encoding defined here as DVI4 is different from those recommendations in two ways 👀

  • The header contains the predicted value rather than the first sample value.
  • IMA ADPCM blocks contain an odd number of samples, since the first sample of a block sits in the header (uncompressed), followed by an even number of compressed samples. DVI4 only has an even number of compressed samples, and it uses the 'predict' word from the header to decode the first sample.

Each packet contains a single DVI block. The profile only defines the 4-bit-per-sample version, while IMA also specifies a 3-bit-per-sample encoding 🍭

The "header" word for each channel has this structure:

  int16  predict;  /* predicted value of first sample
                      from the previous block (L16 format) */
  u_int8 index;    /* current index into stepsize table */
  u_int8 reserved; /* set to zero by sender, ignored by receiver */

Packing of samples for multiple channels is still for further study 🔍

The document "IMA Recommended Practices for Enhancing Digital Audio Compatibility in Multimedia Systems (version 3.0)" has the algorithm description. You can get it from:

Interactive Multimedia Association
48 Maryland Avenue, Suite 202
Annapolis, MD 21401-8011
USA
phone: +1 410 626-1380

4.4.3. G721

G721 is specified in ITU recommendation G.721. Reference implementations for G.721 come as part of the CCITT/ITU-T Software Tool Library (STL), from the ITU General Secretariat, Sales Service, Place du Nations, CH-1211 Geneve 20, Switzerland. Heads up, the library is covered by a license 📜

4.4.4. G722

G722 is specified in ITU-T recommendation G.722, "7 kHz audio-coding within 64 kbit/s" 🎶

G728 is specified in ITU-T recommendation G.728, "Coding of speech at 16 kbit/s using low-delay code excited linear prediction" ⚡

4.4.6. GSM

GSM (group speciale mobile) is the European GSM 06.10 provisional standard for full-rate speech transcoding, prI-ETS 300 036. It's based on RPE/LTP (residual pulse excitation/long term prediction) coding at a rate of 13 kb/s [7,8,9]. You can get the standard from:

ETSI (European Telecommunications Standards Institute)
ETSI Secretariat: B.P.152
F-06561 Valbonne Cedex
France
Phone: +33 92 94 42 00
Fax: +33 93 65 47 16

4.4.7. L8

L8 denotes linear audio data using 8 bits of precision with an offset of 128. That means the most negative signal is encoded as zero 🌸

4.4.8. L16

L16 denotes uncompressed audio data using a 16-bit signed representation, with 65535 equally divided steps between the minimum and maximum signal level, ranging from -32768 to 32767. The value is in two's complement notation and network byte order 🧠

4.4.9. LPC

LPC is an experimental linear predictive encoding contributed by Ron Frederick, Xerox PARC. It's based on an implementation by Ron Zuckerman, Motorola, posted to the Usenet group comp.dsp on June 26, 1992 🕰️

4.4.10. MPA

MPA denotes MPEG-I or MPEG-II audio encapsulated as elementary streams. The encoding is defined in ISO standards ISO/IEC 11172-3 and 13818-3. The encapsulation is in work in progress [10], Section 3. The authors can be contacted at:

Don Hoffman
Sun Microsystems, Inc.
Mail-stop UMPK14-305
2550 Garcia Avenue
Mountain View, California 94043-1100
USA
electronic mail: don.hoffman@eng.sun.com

Sampling rate and channel count are contained in the payload. MPEG-I audio supports sampling rates of 32000, 44100, and 48000 Hz (ISO/IEC 11172-3, section 1.1; "Scope"). MPEG-II additionally supports ISO/IEC 11172-3 Audio...") 🎵

4.4.11. PCMA

PCMA is specified in CCITT/ITU-T recommendation G.711. Audio data is encoded as eight bits per sample, after logarithmic scaling. Code to convert between linear and A-law companded data is available in [6]. For a detailed description, see Jayant and Noll [11] 🧡

4.4.12. PCMU ✨

PCMU is specified in CCITT/ITU-T recommendation G.711 🎀. Audio data gets encoded as eight bits per sample after logarithmic scaling 🔥. Code for converting between linear and mu-law companded data is available in [6] 💅. PCMU is the encoding used for the Internet media type audio/basic 🌸. If you want the deep dive, Jayant and Noll [11] wrote a detailed description 📚🫶

4.4.13. VDVI 🦋

VDVI is basically DVI4's glow-up 💖. It's a variable-rate version that lets speech bit rates bounce anywhere between 10 and 25 kb/s 📈. It's specified for single-channel operation only, so no stereo for this girl 🙈. Here's how the encoding works 👇

                    DVI4 codeword    VDVI bit pattern
                    __________________________________
                                0    00
                                1    010
                                2    1100
                                3    11100
                                4    111100
                                5    1111100
                                6    11111100
                                7    11111110
                                8    10
                                9    011
                               10    1101
                               11    11101
                               12    111101
                               13    1111101
                               14    11111101
                               15    11111111

5. Video ✨

Okay, so these are the video encodings that are currently defined 🎬. Each one has an abbreviated name that's used for identification, and yes, they're all iconic 💜

5.1. CelB 🍓

The CELL-B encoding is a proprietary encoding proposed by Sun Microsystems 💅. The byte stream format is described in work in progress [12]. The author can be contacted at 📬

Michael F. Speer
Sun Microsystems Computer Corporation
2550 Garcia Ave MailStop UMPK14-305
Mountain View, CA 94043
United States
electronic mail: michael.speer@eng.sun.com

5.2. JPEG 📸

The encoding is specified in ISO Standards 10918-1 and 10918-2. The RTP payload format is as specified in work in progress [13]. Further information can be obtained from 🌈

Steven McCanne
Lawrence Berkeley National Laboratory
M/S 46A-1123
One Cyclotron Road
Berkeley, CA 94720
United States
Phone: +1 510 486 7520
electronic mail: mccanne@ee.lbl.gov

5.3. H261 🎥

The encoding is specified in CCITT/ITU-T standard H.261. The packetization and RTP-specific properties are described in work in progress [14]. Further information can be obtained from ✨

Thierry Turletti
Office NE 43-505
Telemedia, Networks and Systems
Laboratory for Computer Science
Massachusetts Institute of Technology
545 Technology Square
Cambridge, MA 02139
United States
electronic mail: turletti@clove.lcs.mit.edu

5.4. MPV 🎞️

MPV designates the use of MPEG-I and MPEG-II video encoding elementary streams, as specified in ISO Standards ISO/IEC 11172 and 13818-2, respectively 🙌. The RTP payload format is as specified in work in progress [10], Section 3. For contact info, see the description of the MPA audio encoding 💁‍♀️

5.5. MP2T 🌟

MP2T designates the use of MPEG-II transport streams, for either audio or video 🎶. The encapsulation is described in work in progress, [10], Section 2. For contact info, see the description of the MPA audio encoding 💁‍♀️

5.6. nv 🧚

The encoding is implemented in the program 'nv', version 4, developed at Xerox PARC by Ron Frederick 👑. Further information is available from the author 🫶

Ron Frederick
Xerox Palo Alto Research Center
3333 Coyote Hill Road
Palo Alto, CA 94304
United States
electronic mail: frederic@parc.xerox.com

6. Payload Type Definitions 👑

Table 2 defines this profile's static payload type values for the PT field of the RTP data header 💅. A new RTP payload format specification may be registered with the IANA by name, and it may also be assigned a static payload type value from the range marked in Section 3 🌸.

Payload type values in the range 96-127 may also be defined dynamically through a conference control protocol, which is beyond the scope of this document 🔮. For example, a session directory could specify that for a given session, payload type 96 indicates PCMU encoding, 8,000 Hz sampling rate, 2 channels 🎀. The payload type range marked 'reserved' has been set aside so that RTCP and RTP packets can be reliably distinguished (see Section "Summary of Protocol Constants" of the RTP protocol specification) 🙈.

An RTP source emits a single RTP payload type at any given time 🦋. Interleaving several RTP payload types in a single RTP session is not allowed, but multiple RTP sessions may be used in parallel to send multiple media 💜. The payload types currently defined in this profile carry either audio or video, but not both 🎧📹. It is allowed to define payload types that combine several media, e.g., audio and video, with appropriate separation in the payload format ✨. Session participants agree on the set of payload types allowed in a given session through mechanisms beyond the scope of this specification 🤝. This set may, for example, be defined by the capabilities of the applications used, negotiated by a conference control protocol, or established by agreement between the human participants 🌈.

Audio applications operating under this profile should, at minimum, be able to send and receive payload types 0 (PCMU) and 5 (DVI4) 🫶. This allows interoperability without format negotiation and successful negotiation with a conference control protocol 🙌.

All current video encodings use a timestamp frequency of 90,000 Hz, the same as the MPEG presentation time stamp frequency ⏰. This frequency yields exact integer timestamp increments for the typical 24 (HDTV), 25 (PAL), and 29.97 (NTSC) and 30 Hz (HDTV) frame rates, and 50, 59.94 and 60 Hz field rates 🎬. While 90 kHz is the recommended rate for future video encodings used within this profile, other rates are possible 🔥. However, it is not sufficient to use the video frame rate (typically between 15 and 30 Hz), because that does not provide adequate resolution for typical synchronization requirements when calculating the RTP timestamp corresponding to the NTP timestamp in an RTCP SR packet [15] 😭. The timestamp resolution must also be sufficient for the jitter estimate contained in the receiver reports 📊.

The standard video encodings and their payload types are listed in Table 2 💖.

7. Port Assignment 🧚

As specified in the RTP protocol definition, RTP data is to be carried on an even UDP port number, and the corresponding RTCP packets are to be carried on the next higher (odd) port number 🦋.

Applications operating under this profile may use any such UDP port pair 🌸. For example, the port pair may be allocated randomly by a session management program 🎲. A single fixed port number pair cannot be required, because multiple applications using this profile are likely to run on the same host 🏠. Some operating systems also do not allow multiple processes to use the same UDP port with different multicast addresses 💀.

      PT         encoding      audio/video    clock rate    channels
                 name          (A/V)          (Hz)          (audio)
      _______________________________________________________________
      0          PCMU          A              8000          1
      1          1016          A              8000          1
      2          G721          A              8000          1
      3          GSM           A              8000          1
      4          unassigned    A              8000          1
      5          DVI4          A              8000          1
      6          DVI4          A              16000         1
      7          LPC           A              8000          1
      8          PCMA          A              8000          1
      9          G722          A              8000          1
      10         L16           A              44100         2
      11         L16           A              44100         1
      12         unassigned    A
      13         unassigned    A
      14         MPA           A              90000        (see text)
      15         G728          A              8000          1
      16--23     unassigned    A
      24         unassigned    V
      25         CelB          V              90000
      26         JPEG          V              90000
      27         unassigned    V
      28         nv            V              90000
      29         unassigned    V
      30         unassigned    V
      31         H261          V              90000
      32         MPV           V              90000
      33         MP2T          AV             90000
      34--71     unassigned    ?
      72--76     reserved      N/A            N/A           N/A
      77--95     unassigned    ?
      96--127    dynamic       ?

Table 2: Payload types (PT) for standard audio and video encodings

However, port numbers 5004 and 5005 have been registered for use with this profile, for those applications that choose to use them as the default pair 🎀. Applications that operate under multiple profiles may use this port pair as an indication to select this profile, if they are not subject to the constraint of the previous paragraph 🔍. Applications need not have a default, and they may require that the port pair be explicitly specified 💅. The particular port numbers were chosen to lie in the range above 5000 to accommodate port number allocation practice within the Unix operating system, where port numbers below 1024 can only be used by privileged processes and port numbers between 1024 and 5000 are automatically assigned by the operating system 🐧✨

8. Bibliography ✨

Okay bestie, time for the reading list 📚 these are the sources the whole spec leans on, so pull them up if you want the deep lore 🤓

[1] Apple Computer, "Audio interchange file format AIFF-C," Aug. 1991. (also ftp://ftp.sgi.com/sgi/aiff-c.9.26.91.ps.Z).

[2] Office of Technology and Standards, "Telecommunications: Analog to digital conversion of radio voice by 4,800 bit/second code excited linear prediction (celp)," Federal Standard FS-1016, GSA, Room 6654; 7th & D Street SW; Washington, DC 20407 (+1-202-708-9205), 1990.

[3] J. P. Campbell, Jr., T. E. Tremain, and V. C. Welch, "The proposed Federal Standard 1016 4800 bps voice coder: CELP," Speech Technology, vol. 5, pp. 58--64, April/May 1990.

[4] J. P. Campbell, Jr., T. E. Tremain, and V. C. Welch, "The federal standard 1016 4800 bps CELP voice coder," Digital Signal Processing, vol. 1, no. 3, pp. 145--155, 1991.

[5] J. P. Campbell, Jr., T. E. Tremain, and V. C. Welch, "The dod 4.8 kbps standard (proposed federal standard 1016)," in Advances in Speech Coding (B. Atal, V. Cuperman, and A. Gersho, eds.), ch. 12, pp. 121--133, Kluwer Academic Publishers, 1991.

[6] IMA Digital Audio Focus and Technical Working Groups, "Recommended practices for enhancing digital audio compatibility in multimedia systems (version 3.00)," tech. rep., Interactive Multimedia Association, Annapolis, Maryland, Oct. 1992.

[7] M. Mouly and M.-B. Pautet, The GSM system for mobile communications Lassay-les-Chateaux, France: Europe Media Duplication, 1993.

[8] J. Degener, "Digital speech compression," Dr. Dobb's Journal, Dec. 1994.

[9] S. M. Redl, M. K. Weber, and M. W. Oliphant, An Introduction to GSM Boston: Artech House, 1995.

[10] D. Hoffman and V. Goyal, "RTP payload format for MPEG1/MPEG2 video," Work in Progress, Internet Engineering Task Force, June 1995.

[11] N. S. Jayant and P. Noll, Digital Coding of Waveforms-- Principles and Applications to Speech and Video Englewood Cliffs, New Jersey: Prentice-Hall, 1984.

[12] M. F. Speer and D. Hoffman, "RTP payload format of CellB video encoding," Work in Progress, Internet Engineering Task Force, Aug. 1995.

[13] W. Fenner, L. Berc, R. Frederick, and S. McCanne, "RTP encapsulation of JPEG-compressed video," Work in Progress, Internet Engineering Task Force, Mar. 1995.

[14] T. Turletti and C. Huitema, "RTP payload format for H.261 video streams," Work in Progress, Internet Engineering Task Force, July 1995.

[15] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: A transport protocol for real-time applications." Work in Progress, Mar. 1995.

9. Security Considerations 🔐

Okay, this one is serious, so don't skip it 😭 Security issues are discussed in section 2. Go read that first, periodt 💅

10. Acknowledgements 🫶

Huge shoutout to Steve Casner 💖 The comments and careful review of Steve Casner are gratefully acknowledged.

11. Author's Address 🎀

Henning Schulzrinne GMD Fokus Hardenbergplatz 2 D-10623 Berlin Germany

EMail: schulzrinne@fokus.gmd.de

These are the spots where you can grab the stuff mentioned above, no cap 🦋

UTF-8

Information on the UCS Transformation Format 8 (UTF-8) is available at

http://www.stonehand.com/unicode/standard/utf8.html

1016

An implementation is available at

ftp://ftp.super.org/pub/speech/celp_3.2a.tar.Z

DVI4

An implementation is available from Jack Jansen at

ftp://ftp.cwi.nl/local/pub/audio/adpcm.shar

G721

An implementation is available at

ftp://gaia.cs.umass.edu/pub/hgschulz/ccitt/ccitt_tools.tar.Z

GSM

A reference implementation was written by Carsten Borman and Jutta Degener (TU Berlin, Germany). It is available at

ftp://ftp.cs.tu-berlin.de/pub/local/kbs/tubmik/gsm/

LPC

An implementation is available at

ftp://parcftp.xerox.com/pub/net-research/lpc.tar.Z

📦 where this page comes from