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author | Thomas Voss <mail@thomasvoss.com> | 2024-11-27 20:54:24 +0100 |
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committer | Thomas Voss <mail@thomasvoss.com> | 2024-11-27 20:54:24 +0100 |
commit | 4bfd864f10b68b71482b35c818559068ef8d5797 (patch) | |
tree | e3989f47a7994642eb325063d46e8f08ffa681dc /doc/rfc/rfc1890.txt | |
parent | ea76e11061bda059ae9f9ad130a9895cc85607db (diff) |
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diff --git a/doc/rfc/rfc1890.txt b/doc/rfc/rfc1890.txt new file mode 100644 index 0000000..80bd170 --- /dev/null +++ b/doc/rfc/rfc1890.txt @@ -0,0 +1,1011 @@ + + + + + + +Network Working Group Audio-Video Transport Working Group +Request for Comments: 1890 H. Schulzrinne +Category: Standards Track GMD Fokus + January 1996 + + + RTP Profile for Audio and Video Conferences with Minimal Control + +Status of this Memo + + This document specifies an Internet standards track protocol for the + Internet community, and requests discussion and suggestions for + improvements. Please refer to the current edition of the "Internet + Official Protocol Standards" (STD 1) for the standardization state + and status of this protocol. Distribution of this memo is unlimited. + +Abstract + + This memo describes a profile for the use of the real-time transport + protocol (RTP), version 2, and the associated control protocol, RTCP, + within audio and video multiparticipant conferences with minimal + control. It provides interpretations of generic fields within the RTP + specification suitable for audio and video conferences. In + particular, this document defines a set of default mappings from + payload type numbers to encodings. + + The document 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. However, the encoding definitions are + independent of the particular transport mechanism used. The + descriptions provide pointers to reference implementations and the + detailed standards. This document is meant as an aid for implementors + of audio, video and other real-time multimedia applications. + +1. Introduction + + This profile defines aspects of RTP left unspecified in the RTP + Version 2 protocol definition (RFC 1889). This profile is intended + for the use within audio and video conferences with minimal session + control. In particular, no support for the negotiation of parameters + or membership control is provided. The profile is expected to be + useful in sessions where no negotiation or membership control are + used (e.g., using the static payload types and the membership + indications provided by RTCP), but this profile may also be useful in + conjunction with a higher-level control protocol. + + + + + + +Schulzrinne Standards Track [Page 1] + +RFC 1890 AV Profile January 1996 + + + Use of this profile occurs by use of the appropriate applications; + there is no explicit indication by port number, protocol identifier + or the like. + + 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" + enumerates a number of items that can be specified or modified in a + profile. This section addresses these items. Generally, this profile + follows 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. Thus, applications should not assume that the RTP + header X bit is always zero and 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. + + + + +Schulzrinne Standards Track [Page 2] + +RFC 1890 AV Profile January 1996 + + + String-to-key mapping: A user-provided string ("pass phrase") is + hashed with the MD5 algorithm to a 16-octet digest. An n-bit key + is extracted from the digest by taking the first n bits from the + digest. If several keys are needed with a total length of 128 + bits or less (as for triple DES), they are extracted in order + from that digest. The octet ordering is specified in RFC 1423, + Section 2.2. (Note that some DES implementations require that + the 56-bit key be expanded into 8 octets by inserting an odd + parity bit in the most significant bit of the octet to go with + each 7 bits of the key.) + + It is suggested that pass phrases are restricted to ASCII letters, + digits, the hyphen, and white space to reduce the the chance of + transcription errors when conveying keys by phone, fax, telex or + email. + + The 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 any additional + identifiers registered with IANA. If no slash is present, DES-CBC is + assumed as default. The encryption algorithm specifier is case + sensitive. + + The pass phrase typed by the user is transformed to a canonical form + before applying the hash algorithm. For that purpose, we define + return, tab, or vertical tab as well as all characters contained in + the Unicode space characters table. The transformation consists of + the following steps: (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 require 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 + by 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 applying the transformation) should be + enforced by the application, while applications must allow up to 256 + characters of input. + + Underlying protocol: The profile specifies the use of RTP over + unicast and multicast UDP. (This does not preclude the use of + these definitions when RTP is carried by other lower-layer + protocols.) + + Transport mapping: The standard mapping of RTP and RTCP to + transport-level addresses is used. + + + + +Schulzrinne Standards Track [Page 3] + +RFC 1890 AV Profile January 1996 + + + Encapsulation: No encapsulation of RTP packets is specified. + +3. Registering Payload Types + + This profile defines a set of standard encodings and their payload + types when used within RTP. Other encodings and their payload types + are to be registered with the Internet Assigned Numbers Authority + (IANA). When registering a new encoding/payload type, the following + information should be provided: + + o name and description of encoding, in particular the RTP + timestamp clock rate; the names defined here are 3 or 4 + characters long to allow a compact representation if needed; + + o indication of 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); + + o any operating parameters or profiles; + + o a reference to a further description, if available, for + example (in order of preference) an RFC, a published paper, a + patent filing, a technical report, documented source code or a + computer manual; + + o for proprietary encodings, contact information (postal and + email address); + + o the payload type value for this profile, if necessary (see + below). + + Note that not all encodings to be used by RTP need to be assigned a + static payload type. Non-RTP means beyond the scope of this memo + (such as directory services or invitation protocols) may be used to + establish a dynamic mapping between a payload type drawn from the + range 96-127 and an encoding. For implementor convenience, this + profile contains descriptions of encodings which do not currently + have a static payload type assigned to them. + + The available payload type space is relatively small. Thus, new + static payload types are assigned only if the following conditions + are met: + + o The encoding is of interest to the Internet community at + large. + + + + + +Schulzrinne Standards Track [Page 4] + +RFC 1890 AV Profile January 1996 + + + o It offers benefits compared to existing encodings and/or is + required for interoperation with existing, widely deployed + conferencing or multimedia systems. + + o The description is sufficient to build a decoder. + +4. Audio + +4.1 Encoding-Independent Recommendations + + For applications which send no packets during silence, the first + packet of a talkspurt (first packet after a silence period) is + distinguished by setting the marker bit in the RTP data header. + Applications without silence suppression set the bit to zero. + + The RTP clock rate used for generating the RTP timestamp is + independent of the number of channels and the encoding; it equals the + number of sampling periods per second. For N-channel encodings, each + sampling period (say, 1/8000 of a second) generates N samples. (This + terminology is standard, but somewhat confusing, as the total number + of samples generated per second is then the sampling rate times the + channel count.) + + If multiple audio channels are used, channels are numbered left-to- + right, starting at one. In RTP audio packets, information from + lower-numbered channels precedes that from higher-numbered channels. + For more than two channels, the convention followed by the AIFF-C + audio interchange format should be followed [1], using the following + notation: + + l left + r right + c center + S surround + F front + R rear + + + + 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 + + + +Schulzrinne Standards Track [Page 5] + +RFC 1890 AV Profile January 1996 + + + Samples for all channels belonging to a single sampling instant must + be within the same packet. The interleaving of samples from different + channels depends on the encoding. General guidelines are given in + Section 4.2 and 4.3. + + The sampling frequency should be drawn from the set: 8000, 11025, + 16000, 22050, 24000, 32000, 44100 and 48000 Hz. (The Apple Macintosh + computers have native sample rates of 22254.54 and 11127.27, which + can be converted to 22050 and 11025 with acceptable quality by + dropping 4 or 2 samples in a 20 ms frame.) However, most audio + encodings are defined for a more restricted set of sampling + frequencies. Receivers should be prepared to accept multi-channel + audio, but may choose to only play a single channel. + + The following recommendations are default operating parameters. + Applications should be prepared to handle other values. The ranges + given are meant to give guidance to application writers, allowing a + set of applications conforming to these guidelines to interoperate + without additional negotiation. These guidelines are not intended to + restrict operating parameters for applications that can negotiate a + set of interoperable parameters, e.g., through a conference control + protocol. + + For packetized audio, the default packetization interval should have + a duration of 20 ms, unless otherwise noted when describing the + encoding. The packetization interval determines the minimum end-to- + end delay; longer packets introduce less header overhead but higher + delay and make packet loss more noticeable. For non-interactive + applications such as lectures or links with severe bandwidth + constraints, a higher packetization delay may be appropriate. A + receiver should accept packets representing between 0 and 200 ms of + audio data. This restriction allows reasonable buffer sizing 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. Within the compressed audio data, codes for + individual samples may span octet boundaries. An RTP audio packet may + contain any number of audio samples, subject to the constraint that + the number of bits per sample times the number of samples per packet + yields an integral octet count. Fractional encodings produce less + than one octet per sample. + + The duration of an audio packet is determined by the number of + samples in the packet. + + + + + +Schulzrinne Standards Track [Page 6] + +RFC 1890 AV Profile January 1996 + + + For sample-based encodings producing one or more octets per sample, + samples from different channels sampled at the same sampling instant + are packed in consecutive octets. For example, for a two-channel + encoding, the octet sequence is (left channel, first sample), (right + channel, first sample), (left channel, second sample), (right + channel, second sample), .... For multi-octet encodings, octets are + transmitted in network byte order (i.e., most significant octet + first). + + The packing of sample-based encodings producing less than one octet + per sample is encoding-specific. + +4.3 Guidelines for Frame-Based Audio Encodings + + Frame-based encodings encode a fixed-length block of audio into + another block of compressed data, typically also of fixed length. For + frame-based encodings, the sender may choose to combine several such + frames into a single message. The receiver can tell the number of + frames contained in a message since the frame duration is defined as + part of the encoding. + + For frame-based codecs, the channel order is defined for the whole + block. That is, for two-channel audio, right and left samples are + coded independently, with the encoded frame for the left channel + preceding that 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 the frame-oriented codecs is given, there is no need to use + a separate designation for the same encoding, but with different + number of frames per packet. + + + + + + + + + + + + + + + + + + + + +Schulzrinne Standards Track [Page 7] + +RFC 1890 AV Profile January 1996 + + +4.4 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 standard audio encodings are shown in Table 1 + and their payload types are listed in Table 2. + +4.4.1 1016 + + Encoding 1016 is a frame based encoding using code-excited linear + prediction (CELP) and is 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 (on DOS diskettes, but configured to compile on Sun SPARC + stations) 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. A specification titled "DVI ADPCM Wave Type" can also be found + in the Microsoft Developer Network Development Library CD ROM + published quarterly by Microsoft. The relevant section is found 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 differs in two + respects from these recommendations: + + + + + +Schulzrinne Standards Track [Page 8] + +RFC 1890 AV Profile January 1996 + + + o The header contains the predicted value rather than the first + sample value. + + o IMA ADPCM blocks contain odd number of samples, since the + first sample of a block is contained just in the header + (uncompressed), followed by an even number of compressed + samples. DVI4 has an even number of compressed samples only, + using 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 the following 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 for further study. + + The document, "IMA Recommended Practices for Enhancing Digital Audio + Compatibility in Multimedia Systems (version 3.0)", contains the + algorithm description. It is available 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 are available 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. 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". + + + +Schulzrinne Standards Track [Page 9] + +RFC 1890 AV Profile January 1996 + + +4.4.6 GSM + + GSM (group speciale mobile) denotes the European GSM 06.10 + provisional standard for full-rate speech transcoding, prI-ETS 300 + 036, which is based on RPE/LTP (residual pulse excitation/long term + prediction) coding at a rate of 13 kb/s [7,8,9]. The standard can be + obtained 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 is, the most negative signal is encoded as zero. + +4.4.8 L16 + + L16 denotes uncompressed audio data, using 16-bit signed + representation with 65535 equally divided steps between minimum and + maximum signal level, ranging from -32768 to 32767. The value is + represented in two's complement notation and network byte order. + +4.4.9 LPC + + LPC designates an experimental linear predictive encoding contributed + by Ron Frederick, Xerox PARC, which is based on an implementation + written 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 specified 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 + + + +Schulzrinne Standards Track [Page 10] + +RFC 1890 AV Profile January 1996 + + + 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]. + A detailed description is given by Jayant and Noll [11]. + +4.4.12 PCMU + + PCMU 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 mu-law companded data is available in [6]. + PCMU is the encoding used for the Internet media type audio/basic. A + detailed description is given by Jayant and Noll [11]. + +4.4.13 VDVI + + VDVI is a variable-rate version of DVI4, yielding speech bit rates of + between 10 and 25 kb/s. It is specified for single-channel operation + only. It uses the following encoding: + + 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 + + + + + + + +Schulzrinne Standards Track [Page 11] + +RFC 1890 AV Profile January 1996 + + +5. Video + + The following video encodings are currently defined, with their + abbreviated names used for identification: + +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 + + + +Schulzrinne Standards Track [Page 12] + +RFC 1890 AV Profile January 1996 + + +5.4 MPV + + MPV designates the use 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. See the description of the MPA audio + encoding for contact information. + +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. See the description of the MPA audio encoding for + contact information. + +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 may also be assigned a + static payload type value from the range marked in Section 3. + + In addition, payload type values in the range 96-127 may 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; the + interleaving of 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 + + + +Schulzrinne Standards Track [Page 13] + +RFC 1890 AV Profile January 1996 + + + profile carry either audio or video, but not both. However, 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 through mechanisms beyond the scope of + this specification on the set of payload types allowed in a given + session. 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 negotation 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, and there are some operating systems that do + not allow multiple processes to use the same UDP port with different + multicast addresses. + + + + + + + + +Schulzrinne Standards Track [Page 14] + +RFC 1890 AV Profile January 1996 + + + 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 may require that the port + pair be explicitly specified. The particular port numbers were chosen + to lie in the range above 5000 to accomodate 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 + + + +Schulzrinne Standards Track [Page 15] + +RFC 1890 AV Profile January 1996 + + + system. + +8. Bibliography + + [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. + + + +Schulzrinne Standards Track [Page 16] + +RFC 1890 AV Profile January 1996 + + + [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 + + Security issues are discussed in section 2. + +10. Acknowledgements + + 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 + + + + + + + + + + + + + + + + + +Schulzrinne Standards Track [Page 17] + +RFC 1890 AV Profile January 1996 + + + Current Locations of Related Resources + + + 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 + + + + + + + +Schulzrinne Standards Track [Page 18] + |