RTP Payload Format for mU-law EMbedded Codec for Low-delay IP Communication (UEMCLIP) Speech Codec
The information below is for an old version of the document that is already published as an RFC.
This is an older version of an Internet-Draft that was ultimately published as RFC 5686.
|Authors||Hitoshi Ohmuro , Yusuke Hiwasaki|
|Last updated||2015-10-14 (Latest revision 2009-05-15)|
|RFC stream||Internet Engineering Task Force (IETF)|
|Additional resources||Mailing list discussion|
|IESG||IESG state||RFC 5686 (Proposed Standard)|
|Responsible AD||Cullen Fluffy Jennings|
|Send notices to||(None)|
Audio/Video Transport Y. Hiwasaki Internet-Draft H. Ohmuro Intended status: Standards Track NTT Corporation Expires: November 15, 2009 May 14, 2009 RTP payload format for mU-law EMbedded Codec for Low-delay IP communication (UEMCLIP) speech codec draft-ietf-avt-rtp-uemclip-06 Status of this Memo This Internet-Draft is submitted to IETF in full conformance with the provisions of BCP 78 and BCP 79. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF), its areas, and its working groups. Note that other groups may also distribute working documents as Internet- Drafts. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress." The list of current Internet-Drafts can be accessed at http://www.ietf.org/ietf/1id-abstracts.txt. The list of Internet-Draft Shadow Directories can be accessed at http://www.ietf.org/shadow.html. This Internet-Draft will expire on November 15, 2009. Copyright Notice Copyright (c) 2009 IETF Trust and the persons identified as the document authors. All rights reserved. This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents in effect on the date of publication of this document (http://trustee.ietf.org/license-info). Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Hiwasaki & Ohmuro Expires November 15, 2009 [Page 1] Internet-Draft RTP Payload Format for UEMCLIP May 2009 Abstract This document describes the RTP payload format of a mU-law EMbedded Coder for Low-delay IP communication (UEMCLIP), an enhanced speech codec of ITU-T G.711. The bitstream has a scalable structure with an embedded u-law bitstream, also known as PCMU, thus providing a handy transcoding operation between narrowband and wideband speech. Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3 1.1. Terminology . . . . . . . . . . . . . . . . . . . . . . . 3 2. Media Format Background . . . . . . . . . . . . . . . . . . . 4 3. Payload Format . . . . . . . . . . . . . . . . . . . . . . . . 6 3.1. RTP Header Usage . . . . . . . . . . . . . . . . . . . . . 6 3.2. Multiple frames in an RTP packet . . . . . . . . . . . . . 7 3.3. Payload Data . . . . . . . . . . . . . . . . . . . . . . . 8 3.3.1. Main Header . . . . . . . . . . . . . . . . . . . . . 8 3.3.2. Sub-layer . . . . . . . . . . . . . . . . . . . . . . 11 4. Transcoding between UEMCLIP and G.711 . . . . . . . . . . . . 13 5. Congestion Control Considerations . . . . . . . . . . . . . . 14 6. Payload Format Parameters . . . . . . . . . . . . . . . . . . 15 6.1. Media type registration . . . . . . . . . . . . . . . . . 15 6.2. Mapping to SDP Parameters . . . . . . . . . . . . . . . . 16 6.2.1. Mode specification . . . . . . . . . . . . . . . . . . 17 6.3. Offer-Answer Model Considerations . . . . . . . . . . . . 17 6.3.1. Offer-Answer Guidelines . . . . . . . . . . . . . . . 18 6.3.2. Examples . . . . . . . . . . . . . . . . . . . . . . . 18 7. Security Considerations . . . . . . . . . . . . . . . . . . . 21 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 22 9. References . . . . . . . . . . . . . . . . . . . . . . . . . . 23 9.1. Normative References . . . . . . . . . . . . . . . . . . . 23 9.2. Informative References . . . . . . . . . . . . . . . . . . 23 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 24 Hiwasaki & Ohmuro Expires November 15, 2009 [Page 2] Internet-Draft RTP Payload Format for UEMCLIP May 2009 1. Introduction This document specifies the payload format for sending "mU-law EMbedded Coder for Low-delay IP communication" (UEMCLIP) encoded speech using the Real-time Transport Protocol (RTP) [RFC3550]. UEMCLIP is a proprietary codec which enhances u-law ITU-T G.711 [ITU-T G.711], and designed to help the market for smooth transition towards the forthcoming wideband communication environment while achieving a very little media transcoding load with the existing terminals, in which the implementation of G.711 is mandatory. It should be noted that, generally speaking, codecs are negotiated and changed using an SDP exchange. Also, [RFC3550] defines general RTP mixer and translator models, where media transcoding may not take place at the node. For those cases, the design concept of the embedded structure is not useful. However, there are other cases when costly transcoding is unavoidable in commonly deployed types of Multi-point Control Units (MCUs) which terminates media and RTCP packets [RFC5117], and when narrowband and wideband terminals co- exist. This embedded bitstream structure can reduce the media transcoding to a simple bitstream truncation. The background and the basic idea of the media format is described in Section 2. The details of the payload format are given in Section 3. The transcoding issues with G.711 are discussed in Section 4, and the considerations for congestion control are in Section 5. In Section 6, the payload format parameters for a media type registration for UEMCLIP RTP payload format and SDP mappings are provided. The security considerations and IANA considerations are dealt in Section 7 and Section 8, respectively. 1.1. Terminology 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 [RFC2119]. Hiwasaki & Ohmuro Expires November 15, 2009 [Page 3] Internet-Draft RTP Payload Format for UEMCLIP May 2009 2. Media Format Background UEMCLIP is an enhanced version of u-law ITU-T G.711, otherwise known as PCMU [RFC4856]. It is targeted at Voice over Internet Protocol (VoIP) applications, and its main goal is to provide a wideband communication platform that is highly interoperable with existing terminals equipped with G.711, and to stimulate the market to gradually shift to using wideband communication. In widely deployed multi-point conferencing systems, the packets usually go through RTCP-terminating MCUs, "Topo-RTCP-terminating-MCU" as defined in [RFC5117]. Because the G.711 bitstream is embedded in the bitstream, costly media transcoding can be avoided in this case. This document does not discuss the implementation details of the encoder and decoder, but only describes the bitstream format. Because of its scalable nature, there are a number of sub-bitstreams (sub-layer) in a UEMCLIP bitstream. By choosing appropriate sub- layers, the codec can adapt to the following requirements: o Sampling frequency, o Number of channels, o Speech quality, and o Bit-rate. The UEMCLIP codec operates at 20-ms frame, and includes three sub- coders as shown in Table 1. The core layer is u-law G.711 at 64 kbit/s, and other two are quality and bandwidth enhancement layers with bit-rate of 16 kbit/s each. +-------+---------------------+----------+--------------------------+ | Layer | Description | Bit-rate | Coding algorithm | +-------+---------------------+----------+--------------------------+ | a | G.711 core | 64 | u-law PCM | | | | | | | b | Lower-band | 16 | Time domain block | | | enhancement | | quantization | | | | | | | c | Higher-band | 16 | MDCT block quantization | +-------+---------------------+----------+--------------------------+ Table 1: Sub-layer description Based on these sub-layers, UEMCLIP codec operates in four modes as shown in Table 2. Here, "Ch" is the number of channels and "Fs" is Hiwasaki & Ohmuro Expires November 15, 2009 [Page 4] Internet-Draft RTP Payload Format for UEMCLIP May 2009 the sampling frequency in kHz. It should be noted that the current version only supports single channel operation and there might be future extensions with multi-channel capabilities. The absent Modes 2 and 5 are reserved for possible future extension to 32 kHz sampling modes. As the mode definition is expected to grow, any other modes not defined in this table MUST NOT be used for compatibility and interoperability reasons. +------+----+----+-------+-------+-------+-------------+------------+ | Mode | Ch | Fs | Layer | Layer | Layer | Bit-rate | Total | | | | | a | b | c | w/o headers | bit-rate | | | | | | | | [kbps] | [kbps] | +------+----+----+-------+-------+-------+-------------+------------+ | 0 | 1 | 8 | x | - | - | 64 | 67.2 | | | | | | | | | | | 1 | 1 | 16 | x | - | x | 80 | 84.0 | | | | | | | | | | | 2 | - | - | - | - | - | - | - | | | | | | | | | | | 3 | 1 | 8 | x | x | - | 80 | 84.0 | | | | | | | | | | | 4 | 1 | 16 | x | x | x | 96 | 100.8 | | | | | | | | | | | 5 | - | - | - | - | - | - | - | +------+----+----+-------+-------+-------+-------------+------------+ Table 2: Mode description UEMCLIP bitstream contains internal headers and other side- information apart from the layer data. This results in total bit- rate larger than the sum of the layers shown in the above table. The detail of the internal headers and auxiliary information are described in Section 3.3.1. Defining the sampling frequency and the number of channels does not result in a singular mode, i.e., there can be multiple modes for the same sampling frequency or number of channels. The supported modes would differ between implementations, thus the sender and the receiver must negotiate what mode to use for transmission. Hiwasaki & Ohmuro Expires November 15, 2009 [Page 5] Internet-Draft RTP Payload Format for UEMCLIP May 2009 3. Payload Format As an RTP payload, UEMCLIP bitstream can contain one or more frames as shown in Figure 1. 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 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | RTP Header | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ | | | one or more frames of UEMCLIP | | | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ Figure 1: RTP payload format UEMCLIP bitstream has a scalable structure, thus it is possible to reconstruct the signal by decoding a part of it. A UEMCLIP frame is composed of a main header (MH) followed by one or more (up to three) sub-layers (SL) as shown in Figure 2. +--+-------+//-+ |MH| SL #1 |...| +--+-------+//-+ Figure 2: A UEMCLIP frame (bitstream format) As a sub-layer, the core layer, i.e., "Layer a", MUST always be included. It should be noted that the location of the core layer may or may not immediately follow MH field. The decoder MUST always refer to the layer indices for proper decoding because the order of the sub-layers is arbitrary. The UEMCLIP bitstream does not explicitly include the following information: mode and sampling frequency (Fs). As described before, this information MUST be exchanged while establishing a connection, for example, by means of SDP. 3.1. RTP Header Usage Each RTP packet starts with a fixed RTP header, as explained in [RFC3550]. The following fields of the RTP fixed header used specifically for UEMCLIP streams are emphasized: Hiwasaki & Ohmuro Expires November 15, 2009 [Page 6] Internet-Draft RTP Payload Format for UEMCLIP May 2009 Payload type: The assignment of an RTP payload type for this packet format is outside the scope of this document, however, it is expected that a payload type in the dynamic range shall be assigned. Timestamp: This encodes the sampling instant of the first speech signal sample in the RTP data packet. For UEMCLIP streams, the RTP timestamp MUST advance based on a clock either at 8000 or 16000 (Hz). In cases where the audio sampling rate can change during a session, the RTP timestamp rate MUST equal to the maximum rate (in Hz) given in the mode range (see Section 6.2.1). This implies that the RTP timestamp rate for UEMCLIP payload type MUST NOT change during a session. For example, for a UEMCLIP stream with 8-kHz audio sampling, where a transition to a 16-kHz audio sampling mode is allowed, the RTP time stamp must always advance using 16-kHz clock rate. For a fixed audio sampling mode, the RTP timestamp rate should be either 8 or 16 kHz, depending on the sampling rate. Marker bit: If the codec is used for applications with discontinuous transmission (DTX, or silence compression), the first packet after a silence period during which packets have not been transmitted contiguously SHOULD have the marker bit in the RTP data header set to one. The marker bit in all other packets MUST be zero. Applications without DTX MUST set the marker bit to zero. 3.2. Multiple frames in an RTP packet More than one UEMCLIP frame may be included in a single RTP packet by a sender. However, senders have the following additional restrictions: o A single RTP packet SHOULD NOT include more UEMCLIP frames than will fit in the path MTU. o All frames contained in a single RTP packet MUST be of the same mode. o Frames MUST NOT be split between RTP packets. It is RECOMMENDED that the number of frames contained within an RTP packet be consistent with the application. Since UEMCLIP is designed for telephony application where delay has a great impact on the quality, then fewer frames per packet for lower delay, is preferable. Hiwasaki & Ohmuro Expires November 15, 2009 [Page 7] Internet-Draft RTP Payload Format for UEMCLIP May 2009 3.3. Payload Data In a UEMCLIP bitstream, all numbers are encoded in a network byte- order. 3.3.1. Main Header The main header (MH) is placed at the top of a frame and has size of 6 bytes. The content of the main header is shown in Figure 3. 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 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | MX | PC | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | PC(cont'd) | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ Figure 3: UEMCLIP main header format (MH) Mixing information (MX): 8 bits Mixing information field. This field is only relevant when Topo- RTCP-terminating-MCUs are utilized to interpret these fields. See Section 184.108.40.206 for details of the fields. Packet-loss Concealment information (PC): 40 bits Packet-loss concealment (PLC) information field. See Section 220.127.116.11. 18.104.22.168. Mixing information field 0 1 2 3 4 5 6 7 +-+-+-+-+-+-+-+-+ |C|R|V| PW1 | |1|1|1| | +-+-+-+-+-+-+-+-+ Figure 4: Mixing information field (MX) Check bit #1 (C1): 1 bit Validity flag of V1 and PW1. This bit being "1" indicates that both parameters are valid, and "0" indicates that the parameters should be ignored. If any of these parameters is invalid, this bit should be set to "0". This flag is mainly intended for a UEMCLIP-conscious Topo-RTCP-terminating-MCU. This flag should be Hiwasaki & Ohmuro Expires November 15, 2009 [Page 8] Internet-Draft RTP Payload Format for UEMCLIP May 2009 set to "0" in case of upward transcoding from G.711 (See Section 4). Reserved bit #1 (R1): 1 bit This bit should be ignored. The default of this bit is 0. VAD flag #1 (V1): 1 bit Voice activity detection flag of the current frame, designed to be used for MCU operations. This flag being "1" indicates that the frame is an active (voice) segment, and "0" indicates that it is an inactive (non-voice) or a silent segment. This flag is specifically designed for mixing information. DTX judgment based this flag is not recommended. Power #1 (PW1): 5 bits Signal power code of the current frame. The code is obtained by calculating a root mean square (RMS) of "Layer a" and encoding this RMS using G.711 u-law [ITU-T G.711]. Denoting the encoded RMS as R, then PW1 is obtained by PW1 = ((~R)>>2) & 0x1F, where "~", ">>", "&" are one's complement arithmetic, right SHIFT, and bitwise AND operators, respectively. 22.214.171.124. PLC information field 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 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |C|R2 |V| K |U| P1 |U| P2 | PW2 | |2| |2| |1| |2| | | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | R3 | | | +-+-+-+-+-+-+-+-+ Figure 5: PLC information field (PC) Check bit #2 (C2): 1 bit Validity flag of V2, K, U1, P1, U2, P2, and PW2. If the flag is "1", it means that all these parameters are valid, and "0" means that the parameters should be ignored. If any of these parameters is invalid, this bit should be set to "0". Similarly to C1, this flag should be set to "0" in case of upward transcoding from G.711 (See Section 4). Hiwasaki & Ohmuro Expires November 15, 2009 [Page 9] Internet-Draft RTP Payload Format for UEMCLIP May 2009 Reserved bit #2 (R2): 2 bits These bits should be ignored. The default of these bits are 0. VAD flag #2 (V2): 1 bit Voice activity detection flag of the current frame, designed to be used for packet-loss concealment. This might not be as same as V1 in the mixing information, and might not be synchronous to the marker bit in the RTP header. DTX judgment based this flag is not recommended. Frame indicator (K): 4 bits This value indicates the frame offset of U2, P2 and PW2. Since it is a better idea to carry the speech feature parameters as PLC information in a different frame to maintain the speech quality, this frame offset value gives which frame the parameters are to be associated with. The value ranges between "0" and "15". If the current frame number is N, for example, the value K indicates that U2, P2 and PW2 are associated with frame of N-K. Frame indicator is equal to the difference in the RTP sequence number when one UEMCLIP frame is contained in a single RTP packet. V/UV flag #1 (U1): 1 bit Voiced/Unvoiced signal indicator of the current frame. This flag being "0" indicates that the frame is a voiced signal segment, and "1" indicates that it is an unvoiced signal segment. Pitch lag #1 (P1): 7 bits Pitch code of the current frame. The actual pitch lag is calculated as P1+20 samples in 8-kHz sampling rate. Pitch lag must be 20 <= pitch length <= 120. Codes ranging between "0x65" and "0x7F" are not used. To obtain the pitch lag, any pitch estimation method can be used, such as the one used in G.711 Appendix I [ITU-T G.711 Appendix 1]. V/UV flag #2 (U2): 1 bit Voiced/Unvoiced signal indicator of the offset frame. This flag being "0" indicates that the frame is a voiced signal segment, and "1" indicates that it is an unvoiced signal segment. The offset value is defined as K. Hiwasaki & Ohmuro Expires November 15, 2009 [Page 10] Internet-Draft RTP Payload Format for UEMCLIP May 2009 Pitch lag #2 (P2): 7 bits Pitch code of the offset frame. The offset value is defined as K. The calculation method is identical to "P1", except that it is based on the signal of offset frame. Power #2 (PW2): 8 bits Signal power code of the offset frame. The offset value is defined as K. Reserved bits #3 (R3): 8 bits These bits should be ignored. The default of all bits are "0". 3.3.2. Sub-layer Sub-layer (SL) is a sub-header followed by layer bitstreams, as shown in Figure 6. The sub-header indicates the layer location and the number of bytes. 0 1 2 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 . . . +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+//-+-+-+ |CI |FI |QI |R4 | SB | LD ... | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+//-+-+-+ Figure 6: Sub-layer format (SL) Channel index (CI): 2 bits Indicates the channel number. For all modes given in Table 2, this should be "0". The detail is given in Table 3. Frequency index (FI): 2 bits Indicates the frequency number. "0" means that the layer is in the base frequency band, higher number means that the layer is in respective frequency band. The detail is given in Table 3. Quality index (QI): 2 bits Indicates the quality layer number. "0" means that the layer is in the base layer, and higher number means that the layer is in respective quality layer. The detail is given in Table 3. Hiwasaki & Ohmuro Expires November 15, 2009 [Page 11] Internet-Draft RTP Payload Format for UEMCLIP May 2009 Reserved #4 (R4): 2 bits Not used (reserved). The default value is "0". Sub-layer Size (SB): 8 bits Indicates the byte size of the following sub-layer data. Layer Data (LD): SB*8 bits The actual sub-layer data. For all the layers shown in Table 1, the layer indices are shown in Table 3. +-------+----+----+----+ | Layer | CI | FI | QI | +-------+----+----+----+ | a | 0 | 0 | 0 | | | | | | | b | 0 | 0 | 1 | | | | | | | c | 0 | 1 | 0 | +-------+----+----+----+ Table 3: Layer indices Hiwasaki & Ohmuro Expires November 15, 2009 [Page 12] Internet-Draft RTP Payload Format for UEMCLIP May 2009 4. Transcoding between UEMCLIP and G.711 As given in Section 2, u-law encoded G.711 bitstream (Layer a) is the core layer of a UEMCLIP bitstream, and is always embedded. This means that media transcoding from UEMCLIP bitstream to G.711 does not have to undergo decoding and re-encoding procedures, but simple extraction would suffice. However, this does not apply for the reverse procedure, i.e., transcoding from G.711 to UEMCLIP, because the auxiliary information in the main header (MH) must be assigned separately. It should be noted that this media transcoding is useful for a Media Translator (Topo-Media-Translator) or a Point to Multipoint Using RTCP Terminating MCU (Topo-RTCP-terminating-MCU) in [RFC5117] and all the requirements apply. This means that a transcoding device of this sort MUST rewrite RTCP packets, together with the RTP media packets. The transcoding from UEMCLIP to u-law G.711 can be done easily by finding an appropriate sub-layer. Within a frame, the transcoder should look for a sub-layer with layer index "0x00", and subsequent LD which has size of SB*8 bits (UEMCLIP has a 20-ms frame thus, SB=160) are the actual G.711 bitstream data. It should be noted that transcoder should not always expect the core layer to be located right after the main header. On the other hand, the transcoding from G.711 to UEMCLIP is not entirely straight-forward. Since there are no means to generate enhancement sub-layers, a G.711 bitstream can only be converted to UEMCLIP Mode 0 bitstream. If the original G.711 bitstream is encoded in A-law, it should first be converted to u-law to become the core layer. Because a UEMCLIP frame size is 20 ms, u-law encoded G.711 bitstream MUST be a 160-sample chunk to become a core layer. For the main header contents, when the UEMCLIP encoder is not available, it should follow the following guidelines. o The check bits for mixing and PLC (C1 and C2) are set to 0. o The reserved bits (R1 to R3) in MH are set to respective default values. For the core layer (i.e., u-law G.711 bitstream), it should have the following sub-layer header: o All CI, FI, QI, and R4 MUST be 0. o Sub-layer size (SB) MUST be 160 for 20 ms frame. Hiwasaki & Ohmuro Expires November 15, 2009 [Page 13] Internet-Draft RTP Payload Format for UEMCLIP May 2009 5. Congestion Control Considerations The general congestion control considerations for transporting RTP data also apply to UEMCLIP over RTP [RFC3550] as well as any applicable RTP profile like AVP [RFC3551]. The bandwidth of a UEMCLIP bitstream can be reduced by changing to lower-bit-rate modes. The embedded layer structure of UEMCLIP may help to control congestion, when dynamic mode changing (see Section 6.2.1) is available, and the range of modes is obtained by offer-answer negotiation as given in Section 6.3. It should be noted that this involves proper RTCP handling when the bit-rate is modified in an RTP translator or a mixer [RFC3550]. 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. It should be treated with care because increased delay means reduced quality. Hiwasaki & Ohmuro Expires November 15, 2009 [Page 14] Internet-Draft RTP Payload Format for UEMCLIP May 2009 6. Payload Format Parameters 6.1. Media type registration This registration is done using the template defined in [RFC4288] and following [RFC4855]. Media type name: audio Media subtype name: UEMCLIP Required parameters: Rate: Defines the sampling rate, and MUST be either 8000 or 16000. See Section 6.2.1 "Mode specification" of RFC xxxx (this RFC) for details. Optional parameters: ptime: See RFC 4566 [RFC4566]. maxptime: See RFC 4566 [RFC4566]. mode: Indicates the range of dynamically changeable modes during a session. Possible values are comma separated list of modes from the supported mode set: 0, 1, 3, and 4. If only one mode is specified, it means that the mode must not be changed during the session. When not specified, the mode transmission defaults to a singular mode as specified in Table 4. See Section 6.2.1 "Mode specification" of RFC xxxx (this RFC) for details. Encoding considerations: This media type is framed and contains binary data. See Section 4.8 of RFC 4288. Security considerations: See Section 7 "Security Considerations" of RFC xxxx (this RFC). Interoperability considerations: This media may be readily transcoded to u-law encoded ITU-T G.711. See Section 4 "Transcoding between UEMCLIP and G.711" of RFC xxxx (this RFC). Published specification: RFC xxxx (This RFC) Applications that use this media type: Audio and video streaming and conferencing tools. Hiwasaki & Ohmuro Expires November 15, 2009 [Page 15] Internet-Draft RTP Payload Format for UEMCLIP May 2009 Additional information: None Intended usage: COMMON Restrictions on usage: This media type depends on RTP framing, and hence is only defined for transfer via RTP. Person & email address to contact for further information: Yusuke Hiwasaki <email@example.com> Author: Author: Yusuke Hiwasaki Change Controller: IETF Audio/Video Transport Working Group delegated from the IESG 6.2. Mapping to SDP Parameters The media types audio/UEMCLIP are mapped to fields in the Session Description Protocol (SDP) [RFC4566] as follows: Media name: The "m=" line of SDP MUST be audio. Encoding name: Registered media subtype name should be used for the "a=rtpmap" line. Sampling Frequency: Depending on the mode, clock rate (sampling frequency) specified in "a=rtpmap" MUST be selected from the ones defined in Table 2. See Section 6.2.1 for details. Encoding parameters: Since this is an audio stream, the encoding parameters indicate the number of audio channels, and this SHOULD default to "1", as selected from the ones defined in Table 2. This is OPTIONAL. Packet time: A frame length of any UEMCLIP is 20 ms, thus the argument of "a=ptime" SHOULD be a multiple of "20". When not listed in SDP, it should also default to the minimum size: "20". UMECLIP specific: Any description specific to UEMCLIP are defined in the Format Specification Parameters ("a=fmtp"). Each parameter MUST be separated with ";", and if any attributes (value) exists, it MUST be defined with "=". For compatibility reasons, any application/terminal MUST ignore any parameters that it does not understand. This is to ensure the upper-compatibility with parameters added in future enhancements. The mode specification should be made here (see Section 6.2.1). Hiwasaki & Ohmuro Expires November 15, 2009 [Page 16] Internet-Draft RTP Payload Format for UEMCLIP May 2009 6.2.1. Mode specification Since UEMCLIP codec can operate in number of modes (bit-rates), it is desirable to specify the range of modes that an encoder or a decoder can operate at. When exchanging SDP messages, an offerer should specify all possible combination of mode numbers as arguments to "mode=" in "a=fmtp" line, delimited by commas ",". In case of specifying multiple modes, those SHOULD appear in the descending priority order. Although UEMCLIP decoders SHOULD accept bitstreams in any modes, an implementation may fail to adopt to the dynamic mode changes during a session. For this reason, an application may choose to operate either with one fixed mode or with multiple modes that can be dynamically changed. If the mode is to be fixed and changes are not allowed, this can be indicated by specifying a single mode per payload type. The mode numbers that can be specified in a payload type as arguments to "mode" are restricted by a combination of a clock rate and a number of audio channels. This is because SDP binds a payload type to a combination of a sampling frequency and a number of audio channels. Table 4 gives selectable mode numbers that attributed with clock rates. When mode specifications are not given at all, a payload type MUST default to a single mode using the default value specified in this table. +------------+----------+------------------+--------------+ | Clock rate | Channels | Selectable modes | Default mode | +------------+----------+------------------+--------------+ | 8000 | 1 | 0,3 | 0 | | | | | | | 16000 | 1 | 0,1,3,4 | 1 | +------------+----------+------------------+--------------+ Table 4: Default modes It should be noted that a mode attributed with a larger sampling frequency (Fs) is not used in conjunction with smaller clock rates specified in "a=rtpmap". This means that Modes 0 and 3 can be specified in a payload type having clock rate of both 8000 and 16000 in "a=rtpmap", but Modes 1 and 4 cannot be specified with one having clock rate of 8000. 6.3. Offer-Answer Model Considerations Hiwasaki & Ohmuro Expires November 15, 2009 [Page 17] Internet-Draft RTP Payload Format for UEMCLIP May 2009 6.3.1. Offer-Answer Guidelines The procedures related to exchanging SDP messages MUST follow [RFC3264]. The following is a detalied list on the semantics of using the UEMCLIP payload format in an offer-answer exchange. o An offerer SHOULD offer every possible combination of UEMCLIP payload type it can handle, i.e., sampling frequency, channel number, and fmtp parameters, in a preferred order. When the transmission bandwidth is restricted, it MUST be offered in accordance to the restriction. o When multiple UEMCLIP payload types are offered, it is RECOMMENDED that the answerer selects a single UEMCLIP payload type and answers it back. o In a UEMCLIP payload type, an answerer MUST answer back suitable mode number(s) as a subset of what has been offered. This means that there is a symmetry assumption on sent and received streams, and offerer MUST NOT send in modes that it does not offer. o In an offering/answering SDP, any fmtp parameters which are not known MUST be ignored. If any unknown/undefined parameters should be offered, an answerer MUST delete the entry from the answer message. o A receiver of an SDP message MUST only use specified payload types and modes. When a mode specification is missing, i.e., a mode is not specified at all, the session MUST default to one single mode without mode changes during a session. For this case, the default mode values, as shown in Table 4, MUST be used based on the sampling frequency and number of channels. This table must be looked up only when there are no mode specifications, thus the offerer/answerer MUST NOT assume that the default modes are always available when it is not in the specified list of modes. o When an offered condition does not fit an answerer's capabilities, it naturally MUST NOT answer any of the conditions, and session MAY proceed to re-INVITE, if possible. If a condition (mode) is decided upon, an offerer and an answerer MUST transmit on this condition. 6.3.2. Examples When an offerer indicates that he/she wishes to dynamically switch between modes (0,1,3, and 4) during a session, an example of an offered SDP can be: Hiwasaki & Ohmuro Expires November 15, 2009 [Page 18] Internet-Draft RTP Payload Format for UEMCLIP May 2009 v=0 o=john 51050101 51050101 IN IP4 offhost.example.com s=- c=IN IP4 offhost.example.com t=0 0 m=audio 5004 RTP/AVP 96 a=rtpmap:96 UEMCLIP/16000/1 a=fmtp:96 mode=4,1,3,0 It should be noted that the listed modes appears in the offerer's preference. When an answerer can only operate in Modes 1 and 0 but can dynamically switch between those modes during a session, an answerer MUST delete the entries of Mode 3 and 4, and answer back as: v=0 o=lena 549947322 549947322 IN IP4 anshost.example.org s=- c=IN IP4 anshost.example.org t=0 0 m=audio 5004 RTP/AVP 96 a=rtpmap:96 UEMCLIP/16000/1 a=fmtp:96 mode=1,0 As a result, both would start communicating in either Mode 1 or 0, and can dynamically switch between those modes during the session. On the other hand, when the answerer is capable of communicating either in Modes 1 or 0, and cannot switch between modes during a session, an example of such answer is as follows: v=0 o=lena 549947322 549947322 IN IP4 anshost.example.org s=- c=IN IP4 anshost.example.org t=0 0 m=audio 5004 RTP/AVP 96 a=rtpmap:96 UEMCLIP/16000/1 a=fmtp:96 mode=1 As a result, both will start communicating in Mode 1. It should be noted that mode change during this session is not allowed because the answerer responded with a single mode, and answerer selected Mode 1 above Mode 0 according to the offered order. If an offerer does not want a mode change during a session but is capable of receiving either Modes 4 or 1 bitstreams, the SDP should Hiwasaki & Ohmuro Expires November 15, 2009 [Page 19] Internet-Draft RTP Payload Format for UEMCLIP May 2009 somewhat look like: v=0 o=john 51050101 51050101 IN IP4 offhost.example.com s=- c=IN IP4 offhost.example.com t=0 0 m=audio 5004 RTP/AVP 96 97 a=rtpmap:96 UEMCLIP/16000/1 a=fmtp:96 mode=4 a=rtpmap:97 UEMCLIP/16000/1 a=fmtp:97 mode=1 and if the answerer prefers to communicate in Mode 1, an answer would be: v=0 o=lena 549947322 549947322 IN IP4 anshost.example.org s=- c=IN IP4 anshost.example.org t=0 0 m=audio 5004 RTP/AVP 97 a=rtpmap:97 UEMCLIP/16000/1 a=fmtp:97 mode=1 Please note that it is RECOMMENDED to select a single UEMCLIP payload type for answers. The "ptime" attribute is used to denote the desired packetization interval. When not specified, it SHOULD default to 20. Since UEMCLIP uses 20 msec frames, ptime values of multiples of 20 imply multiple frames per packet. In the example below, the ptime is set to 60, and this means that offerer wants to receive 3 frames in each packet. v=0 o=kosuke 2890844730 2890844730 IN IP4 anotherhost.example.com s=- c=IN IP4 anotherhost.example.com t=0 0 m=audio 5004 RTP/AVP 96 a=ptime:60 a=rtpmap:96 UEMCLIP/16000/1 When mode specification is not present, it should default to a fixed mode, and in this case, Mode 1 (see Section 6.2.1). Hiwasaki & Ohmuro Expires November 15, 2009 [Page 20] Internet-Draft RTP Payload Format for UEMCLIP May 2009 7. 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 any appropriate profiles. This implies that confidentiality of the media streams is achieved by encryption unless the applicable profile specifies other means. A potential denial-of-service threat exists for data encoding using compression techniques that have non-uniform receiver-end computational load. The attacker can inject pathological datagrams into the stream that are complex to decode and cause the receiver output to become overloaded. However, UEMCLIP covered in this document do not exhibit any significant non-uniformity. Another potential threats are memory attacks by illegal layer indices or byte numbers. The implementor of the decoder should always be aware that the indicated numbers may be corrupted and does not point to the right sub-layer and may force reading beyond the bitstream boundaries. It is advised that a decoder implementation rejects layers such indices. Hiwasaki & Ohmuro Expires November 15, 2009 [Page 21] Internet-Draft RTP Payload Format for UEMCLIP May 2009 8. IANA Considerations It is requested that one new media subtype (audio/UEMCLIP) is registered by IANA. For details, see Section 6.1. Hiwasaki & Ohmuro Expires November 15, 2009 [Page 22] Internet-Draft RTP Payload Format for UEMCLIP May 2009 9. References 9.1. 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. [RFC4288] Freed, N. and J. Klensin, "Media Type Specifications and Registration Procedures", BCP 13, RFC 4288, December 2005. [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session Description Protocol", RFC 4566, July 2006. [RFC4855] Casner, S., "Media Type Registration of RTP Payload Formats", RFC 4855, February 2007. [RFC4856] Casner, S., "Media Type Registration of Payload Formats in the RTP Profile for Audio and Video Conferences", RFC 4856, February 2007. [ITU-T G.711] International Telecommunications Union, "Pulse code modulcation (PCM) of Voice Frequencies", ITU- T Recommendation G.711, November 1988. 9.2. Informative References [RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117, January 2008. [ITU-T G.711 Appendix 1] International Telecommunications Union, "Pulse code modulcation (PCM) of Voice Frequencies, Appendix I: A high quality low-complexity algorithm for packet loss concealment with G.711", ITU-T Recommendation G.711 Appendix I, September 1999. Hiwasaki & Ohmuro Expires November 15, 2009 [Page 23] Internet-Draft RTP Payload Format for UEMCLIP May 2009 Authors' Addresses Yusuke Hiwasaki NTT Corporation 3-9-11 Midori-cho, Musashino-shi Tokyo 180-8585 Japan Phone: +81(422)59-4815 Email: firstname.lastname@example.org Hitoshi Ohmuro NTT Corporation 3-9-11 Midori-cho, Musashino-shi Tokyo 180-8585 Japan Phone: +81(422)59-2151 Email: email@example.com Hiwasaki & Ohmuro Expires November 15, 2009 [Page 24]