Request for Comments: 5686
Category: Standards Track
RTP Payload Format for mU-law EMbedded Codec for Low-delay IP
Communication (UEMCLIP) Speech Codec
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.
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.
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Table of Contents
1. Introduction ....................................................2 1.1. Terminology ................................................3 2. Media Format Background .........................................3 3. Payload Format ..................................................5 3.1. RTP Header Usage ...........................................6 3.2. Multiple Frames in an RTP Packet ...........................6 3.3. Payload Data ...............................................7 3.3.1. Main Header .........................................7 3.3.2. Sub-Layer ..........................................10 4. Transcoding between UEMCLIP and G.711 ..........................11 5. Congestion Control Considerations ..............................12 6. Payload Format Parameters ......................................13 6.1. Media Type Registration ...................................13 6.2. Mapping to SDP Parameters .................................14 6.2.1. Mode Specification .................................15 6.3. Offer-Answer Model Considerations .........................16 6.3.1. Offer-Answer Guidelines ............................16 6.3.2. Examples ...........................................17 7. Security Considerations ........................................19 8. IANA Considerations ............................................19 9. References .....................................................19 9.1. Normative References ......................................19 9.2. Informative References ....................................20
This document specifies the payload format for sending UEMCLIP- encoded (mU-law EMbedded Coder for Low-delay IP communication) speech using the Real-time Transport Protocol (RTP) [RFC3550]. UEMCLIP is a proprietary codec that enhances u-law ITU-T G.711 [ITU-T-G.711] and that is designed to help the market for smooth transition towards the forthcoming wideband communication environment while achieving a very small 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 terminate media and RTCP packets [RFC5117], and when narrowband and wideband terminals coexist. 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 Session Description Protocol (SDP) mappings are provided. The security considerations and IANA considerations are dealt with in Section 7 and Section 8, respectively.
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].
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 (RTP Control Protocol) 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:
- Sampling frequency,
- Number of channels,
- Speech quality, and
The UEMCLIP codec operates at a 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, the UEMCLIP codec operates in four modes as shown in Table 2. Here, "Ch" is the number of channels and "Fs" is 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 | | | | | | | | [kbit/s] | [kbit/s] | +------+----+----+-------+-------+-------+-------------+------------+ | 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
The 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.
3. Payload Format
As an RTP payload, the 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
The 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 (SLs) 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:
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 be 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 the 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:
- A single RTP packet SHOULD NOT include more UEMCLIP frames than will fit in the path MTU.
- All frames contained in a single RTP packet MUST be of the same mode.
- 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 applications where delay has a great impact on the quality, then fewer frames per packet for lower delay, is preferable.
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 a 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 188.8.131.52 for details of the fields.
Packet-loss Concealment information (PC): 40 bits
Packet-loss concealment (PLC) information field. See Section 184.108.40.206.
220.127.116.11. 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 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.
18.104.22.168. 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).
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 the 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 with which frame the parameters are to be associated. 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 the frame of N-K. The 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.711Appendix1].
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.
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".
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.
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
4. Transcoding between UEMCLIP and G.711
As given in Section 2, the 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 the 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 a layer index of "0x00", and subsequent LD that has a 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 the 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 straightforward. 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, a 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 these guidelines:
- The check bits for mixing and PLC (C1 and C2) are set to 0.
- 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:
- All CI, FI, QI, and R4 MUST be 0.
- Sub-layer size (SB) MUST be 160 for a 20-ms frame.
5. Congestion Control Considerations
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.
6. Payload Format Parameters
6.1. Media Type Registration
Media type name: audio Media subtype name: UEMCLIP
Rate: Defines the sampling rate, and it MUST be either 8000 or 16000. See Section 6.2.1 "Mode specification" of RFC 5686 (this RFC) for details.
ptime: See RFC 4566 [RFC4566]. maxptime: See RFC 4566 [RFC4566]. mode: Indicates the range of dynamically changeable modes during a session. Possible values are a 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 5686 (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 5686 (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 5686 (this RFC). Published specification: RFC 5686 (this RFC) Applications that use this media type: Audio and video streaming and conferencing tools. 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 <firstname.lastname@example.org> 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 is defined in the Format Specification Parameters ("a=fmtp"). Each parameter MUST be separated with ";", and if any attribute (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).
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 at which an encoder or a decoder can operate. When exchanging SDP messages, an offerer should specify all possible combinations 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 adapt 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 are 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 a clock rate of both 8000 and 16000 in "a=rtpmap", but Modes 1 and 4 cannot be specified with one having a clock rate of 8000.
6.3. Offer-Answer Model Considerations
6.3.1. Offer-Answer Guidelines
The procedures related to exchanging SDP messages MUST follow [RFC3264]. The following is a detailed list on the semantics of using the UEMCLIP payload format in an offer-answer exchange.
- 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.
- When multiple UEMCLIP payload types are offered, it is RECOMMENDED that the answerer select a single UEMCLIP payload type and answer it back.
- 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 the offerer MUST NOT send in modes that it does not offer.
- In an offering/answering SDP, any fmtp parameters that are not known MUST be ignored. If any unknown/undefined parameters should be offered, an answerer MUST delete the entry from the answer message.
- 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.
- When an offered condition does not fit an answerer's capabilities, it naturally MUST NOT answer any of the conditions, and the 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.
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 could be:
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 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-ms 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).
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, the UEMCLIP covered in this document do not exhibit any significant non-uniformity.
Another potential threat is 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 not point to the right sub-layer, and they may force reading beyond the bitstream boundaries. It is advised that a decoder implementation reject layers of such indices.
8. IANA Considerations
One new media subtype (audio/UEMCLIP) has been registered by IANA. For details, see Section 6.1.
9.1. Normative References
International Telecommunications Union, "Pulse code
modulation (PCM) of voice frequencies", ITU-
T Recommendation G.711, November 1988.
[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. [RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117, January 2008.
9.2. Informative References
International Telecommunications Union, "Pulse code modulation (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.
Yusuke Hiwasaki NTT Corporation 3-9-11 Midori-cho, Musashino-shi Tokyo 180-8585 Japan Phone: +81(422)59-4815 EMail: email@example.com Hitoshi Ohmuro NTT Corporation 3-9-11 Midori-cho, Musashino-shi Tokyo 180-8585 Japan Phone: +81(422)59-2151 EMail: firstname.lastname@example.org