RFC 4867
RTP Payload Format and File Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband (AMR-WB) Audio Codecs
Network Working Group J. Sjoberg Request for Comments: 4867 M. Westerlund Obsoletes: 3267 Ericsson Category: Standards Track A. Lakaniemi Nokia Q. Xie Motorola April 2007 RTP Payload Format and File Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband (AMR-WB) Audio Codecs 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. Copyright Notice Copyright (C) The IETF Trust (2007).Abstract
This document specifies a Real-time Transport Protocol (RTP) payload format to be used for Adaptive Multi-Rate (AMR) and Adaptive Multi- Rate Wideband (AMR-WB) encoded speech signals. The payload format is designed to be able to interoperate with existing AMR and AMR-WB transport formats on non-IP networks. In addition, a file format is specified for transport of AMR and AMR-WB speech data in storage mode applications such as email. Two separate media type registrations are included, one for AMR and one for AMR-WB, specifying use of both the RTP payload format and the storage format. This document obsoletes RFC 3267.
Table of Contents
1. Introduction ....................................................4 2. Conventions and Acronyms ........................................4 3. Background on AMR/AMR-WB and Design Principles ..................5 3.1. The Adaptive Multi-Rate (AMR) Speech Codec .................5 3.2. The Adaptive Multi-Rate Wideband (AMR-WB) Speech Codec .....6 3.3. Multi-Rate Encoding and Mode Adaptation ....................6 3.4. Voice Activity Detection and Discontinuous Transmission ....7 3.5. Support for Multi-Channel Session ..........................7 3.6. Unequal Bit-Error Detection and Protection .................8 3.6.1. Applying UEP and UED in an IP Network ...............8 3.7. Robustness against Packet Loss ............................10 3.7.1. Use of Forward Error Correction (FEC) ..............10 3.7.2. Use of Frame Interleaving ..........................12 3.8. Bandwidth-Efficient or Octet-Aligned Mode .................12 3.9. AMR or AMR-WB Speech over IP Scenarios ....................13 4. AMR and AMR-WB RTP Payload Formats .............................15 4.1. RTP Header Usage ..........................................15 4.2. Payload Structure .........................................17 4.3. Bandwidth-Efficient Mode ..................................17 4.3.1. The Payload Header .................................17 4.3.2. The Payload Table of Contents ......................18 4.3.3. Speech Data ........................................20 4.3.4. Algorithm for Forming the Payload ..................21 4.3.5. Payload Examples ...................................21 4.3.5.1. Single-Channel Payload Carrying a Single Frame ..............................21 4.3.5.2. Single-Channel Payload Carrying Multiple Frames ...........................22 4.3.5.3. Multi-Channel Payload Carrying Multiple Frames ...........................23 4.4. Octet-Aligned Mode ........................................25 4.4.1. The Payload Header .................................25 4.4.2. The Payload Table of Contents and Frame CRCs .......26 4.4.2.1. Use of Frame CRC for UED over IP ..........28 4.4.3. Speech Data ........................................30 4.4.4. Methods for Forming the Payload ....................31 4.4.5. Payload Examples ...................................32 4.4.5.1. Basic Single-Channel Payload Carrying Multiple Frames ..................32 4.4.5.2. Two-Channel Payload with CRC, Interleaving, and Robust Sorting ..........32 4.5. Implementation Considerations .............................33 4.5.1. Decoding Validation ................................34 5. AMR and AMR-WB Storage Format ..................................35 5.1. Single-Channel Header .....................................35 5.2. Multi-Channel Header ......................................36
5.3. Speech Frames .............................................37
6. Congestion Control .............................................38
7. Security Considerations ........................................38
7.1. Confidentiality ...........................................39
7.2. Authentication and Integrity ..............................39
8. Payload Format Parameters ......................................39
8.1. AMR Media Type Registration ...............................40
8.2. AMR-WB Media Type Registration ............................44
8.3. Mapping Media Type Parameters into SDP ....................47
8.3.1. Offer-Answer Model Considerations ..................48
8.3.2. Usage of Declarative SDP ...........................50
8.3.3. Examples ...........................................51
9. IANA Considerations ............................................53
10. Changes from RFC 3267 .........................................53
11. Acknowledgements ..............................................55
12. References ....................................................55
12.1. Normative References .....................................55
12.2. Informative References ...................................56
1. Introduction
This document obsoletes RFC 3267 and extends that specification with offer/answer rules. See Section 10 for the changes made to this format in relation to RFC 3267. This document specifies the payload format for packetization of AMR and AMR-WB encoded speech signals into the Real-time Transport Protocol (RTP) [8]. The payload format supports transmission of multiple channels, multiple frames per payload, the use of fast codec mode adaptation, robustness against packet loss and bit errors, and interoperation with existing AMR and AMR-WB transport formats on non-IP networks, as described in Section 3. The payload format itself is specified in Section 4. A related file format is specified in Section 5 for transport of AMR and AMR-WB speech data in storage mode applications such as email. In Section 8, two separate media type registrations are provided, one for AMR and one for AMR-WB. Even though this RTP payload format definition supports the transport of both AMR and AMR-WB speech, it is important to remember that AMR and AMR-WB are two different codecs and they are always handled as different payload types in RTP.2. Conventions and Acronyms
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 RFC 2119 [5]. The following acronyms are used in this document: 3GPP - the Third Generation Partnership Project AMR - Adaptive Multi-Rate (Codec) AMR-WB - Adaptive Multi-Rate Wideband (Codec) CMR - Codec Mode Request CN - Comfort Noise DTX - Discontinuous Transmission ETSI - European Telecommunications Standards Institute FEC - Forward Error Correction SCR - Source Controlled Rate Operation SID - Silence Indicator (the frames containing only CN parameters) VAD - Voice Activity Detection UED - Unequal Error Detection UEP - Unequal Error Protection
The term "frame-block" is used in this document to describe the time-synchronized set of speech frames in a multi-channel AMR or AMR-WB session. In particular, in an N-channel session, a frame- block will contain N speech frames, one from each of the channels, and all N speech frames represents exactly the same time period. The byte order used in this document is network byte order, i.e., the most significant byte first. The bit order is also the most significant bit first. This is presented in all figures as having the most significant bit leftmost on a line and with the lowest number. Some bit fields may wrap over multiple lines in which cases the bits on the first line are more significant than the bits on the next line.3. Background on AMR/AMR-WB and Design Principles
AMR and AMR-WB were originally designed for circuit-switched mobile radio systems. Due to their flexibility and robustness, they are also suitable for other real-time speech communication services over packet-switched networks such as the Internet. Because of the flexibility of these codecs, the behavior in a particular application is controlled by several parameters that select options or specify the acceptable values for a variable. These options and variables are described in general terms at appropriate points in the text of this specification as parameters to be established through out-of-band means. In Section 8, all of the parameters are specified in the form of media subtype registrations for the AMR and AMR-WB encodings. The method used to signal these parameters at session setup or to arrange prior agreement of the participants is beyond the scope of this document; however, Section 8.3 provides a mapping of the parameters into the Session Description Protocol (SDP) [11] for those applications that use SDP.3.1. The Adaptive Multi-Rate (AMR) Speech Codec
The AMR codec was originally developed and standardized by the European Telecommunications Standards Institute (ETSI) for GSM cellular systems. It is now chosen by the Third Generation Partnership Project (3GPP) as the mandatory codec for third generation (3G) cellular systems [1]. The AMR codec is a multi-mode codec that supports eight narrow band speech encoding modes with bit rates between 4.75 and 12.2 kbps. The sampling frequency used in AMR is 8000 Hz and the speech encoding is performed on 20 ms speech frames. Therefore, each encoded AMR speech frame represents 160 samples of the original speech.
Among the eight AMR encoding modes, three are already separately adopted as standards of their own. Particularly, the 6.7 kbps mode is adopted as PDC-EFR [18], the 7.4 kbps mode as IS-641 codec in TDMA [17], and the 12.2 kbps mode as GSM-EFR [16].3.2. The Adaptive Multi-Rate Wideband (AMR-WB) Speech Codec
The Adaptive Multi-Rate Wideband (AMR-WB) speech codec [3] was originally developed by 3GPP to be used in GSM and 3G cellular systems. Similar to AMR, the AMR-WB codec is also a multi-mode speech codec. AMR-WB supports nine wide band speech coding modes with respective bit rates ranging from 6.6 to 23.85 kbps. The sampling frequency used in AMR-WB is 16000 Hz and the speech processing is performed on 20 ms frames. This means that each AMR-WB encoded frame represents 320 speech samples.3.3. Multi-Rate Encoding and Mode Adaptation
The multi-rate encoding (i.e., multi-mode) capability of AMR and AMR-WB is designed for preserving high speech quality under a wide range of transmission conditions. With AMR or AMR-WB, mobile radio systems are able to use available bandwidth as effectively as possible. For example, in GSM it is possible to dynamically adjust the speech encoding rate during a session so as to continuously adapt to the varying transmission conditions by dividing the fixed overall bandwidth between speech data and error protective coding. This enables the best possible trade-off between speech compression rate and error tolerance. To perform mode adaptation, the decoder (speech receiver) needs to signal the encoder (speech sender) the new mode it prefers. This mode change signal is called Codec Mode Request or CMR. Since in most sessions speech is sent in both directions between the two ends, the mode requests from the decoder at one end to the encoder at the other end are piggy-backed over the speech frames in the reverse direction. In other words, there is no out-of-band signaling needed for sending CMRs. Every AMR or AMR-WB codec implementation is required to support all the respective speech coding modes defined by the codec and must be able to handle mode switching to any of the modes at any time. However, some transport systems may impose limitations in the number of modes supported and how often the mode can change due to bandwidth
limitations or other constraints. For this reason, the decoder is allowed to indicate its acceptance of a particular mode or a subset of the defined modes for the session using out-of-band means. For example, the GSM radio link can only use a subset of at most four different modes in a given session. This subset can be any combination of the eight AMR modes for an AMR session or any combination of the nine AMR-WB modes for an AMR-WB session. Moreover, for better interoperability with GSM through a gateway, the decoder is allowed to use out-of-band means to set the minimum number of frames between two mode changes and to limit the mode change among neighboring modes only. Section 8 specifies a set of media type parameters that may be used to signal these mode adaptation controls at session setup.3.4. Voice Activity Detection and Discontinuous Transmission
Both codecs support voice activity detection (VAD) and generation of comfort noise (CN) parameters during silence periods. Hence, the codecs have the option to reduce the number of transmitted bits and packets during silence periods to a minimum. The operation of sending CN parameters at regular intervals during silence periods is usually called discontinuous transmission (DTX) or source controlled rate (SCR) operation. The AMR or AMR-WB frames containing CN parameters are called Silence Indicator (SID) frames. See more details about VAD and DTX functionality in [9] and [10].3.5. Support for Multi-Channel Session
Both the RTP payload format and the storage format defined in this document support multi-channel audio content (e.g., a stereophonic speech session). Although AMR and AMR-WB codecs themselves do not support encoding of multi-channel audio content into a single bit stream, they can be used to separately encode and decode each of the individual channels. To transport (or store) the separately encoded multi-channel content, the speech frames for all channels that are framed and encoded for the same 20 ms periods are logically collected in a frame-block. At the session setup, out-of-band signaling must be used to indicate the number of channels in the session, and the order of the speech frames from different channels in each frame-block. When using SDP for signaling, the number of channels is specified in the rtpmap attribute and the order of channels carried in each frame-block is
implied by the number of channels as specified in Section 4.1 in [12].3.6. Unequal Bit-Error Detection and Protection
The speech bits encoded in each AMR or AMR-WB frame have different perceptual sensitivity to bit errors. This property has been exploited in cellular systems to achieve better voice quality by using unequal error protection and detection (UEP and UED) mechanisms. The UEP/UED mechanisms focus the protection and detection of corrupted bits to the perceptually most sensitive bits in an AMR or AMR-WB frame. In particular, speech bits in an AMR or AMR-WB frame are divided into class A, B, and C, where bits in class A are the most sensitive and bits in class C the least sensitive (see Table 1 below for AMR and [4] for AMR-WB). An AMR or AMR-WB frame is only declared damaged if there are bit errors found in the most sensitive bits, i.e., the class A bits. On the other hand, it is acceptable to have some bit errors in the other bits, i.e., class B and C bits. Class A Total speech Index Mode bits bits ---------------------------------------- 0 AMR 4.75 42 95 1 AMR 5.15 49 103 2 AMR 5.9 55 118 3 AMR 6.7 58 134 4 AMR 7.4 61 148 5 AMR 7.95 75 159 6 AMR 10.2 65 204 7 AMR 12.2 81 244 8 AMR SID 39 39 Table 1. The number of class A bits for the AMR codec Moreover, a damaged frame is still useful for error concealment at the decoder since some of the less sensitive bits can still be used. This approach can improve the speech quality compared to discarding the damaged frame.3.6.1. Applying UEP and UED in an IP Network
To take full advantage of the bit-error robustness of the AMR and AMR-WB codec, the RTP payload format is designed to facilitate UEP/UED in an IP network. It should be noted however that the utilization of UEP and UED discussed below is OPTIONAL.
UEP/UED in an IP network can be achieved by detecting bit errors in class A bits and tolerating bit errors in class B/C bits of the AMR or AMR-WB frame(s) in each RTP payload. Link-layer protocols exist that do not discard packets containing bit errors, e.g., SLIP and some wireless links. With the Internet traffic pattern shifting towards a more multimedia-centric one, more link layers of such nature may emerge in the future. With transport layer support for partial checksums (for example, those supported by UDP-Lite [19]), bit error tolerant AMR and AMR-WB traffic could achieve better performance over these types of links. The relationship between UDP-Lite's partial checksum at the transport layer and the checksum coverage provided by the link-layer frame is described in UDP-Lite specification [19]. There are at least two basic approaches for carrying AMR and AMR-WB traffic over bit error tolerant IP networks: a) Utilizing a partial checksum to cover the IP, transport protocol (e.g., UDP-Lite), RTP and payload headers, and the most important speech bits of the payload. The IP, UDP and RTP headers need to be protected, and it is recommended that at least all class A bits are covered by the checksum. b) Utilizing a partial checksum to only cover the IP, transport protocol, RTP and payload headers, but an AMR or AMR-WB frame CRC to cover the class A bits of each speech frame in the RTP payload. In either approach, at least part of the class B/C bits are left without error-check and thus bit error tolerance is achieved. Note, it is still important that the network designer pays attention to the class B and C residual bit error rate. Though less sensitive to errors than class A bits, class B and C bits are not insignificant, and undetected errors in these bits cause degradation in speech quality. An example of residual error rates considered acceptable for AMR in the Universal Mobile Telecommunications System (UMTS) can be found in [24] and for AMR-WB in [25]. The application interface to the UEP/UED transport protocol (e.g., UDP-Lite) may not provide any control over the link error rate, especially in a gateway scenario. Therefore, it is incumbent upon the designer of a node with a link interface of this type to choose a residual bit error rate that is low enough to support applications such as AMR encoding when transmitting packets of a UEP/UED transport protocol.
Approach 1 is bit efficient, flexible and simple, but comes with two disadvantages, namely, a) bit errors in protected speech bits will cause the payload to be discarded, and b) when transporting multiple AMR or AMR-WB frames in a RTP payload, there is the possibility that a single bit error in protected bits will cause all the frames to be discarded. These disadvantages can be avoided, if needed, with some overhead in the form of a frame-wise CRC (Approach 2). In problem a), the CRC makes it possible to detect bit errors in class A bits and use the frame for error concealment, which gives a small improvement in speech quality. For b), when transporting multiple frames in a payload, the CRCs remove the possibility that a single bit error in a class A bit will cause all the frames to be discarded. Avoiding that improves the speech quality when transporting multiple AMR or AMR-WB frames over links subject to bit errors. The choice between the above two approaches must be made based on the available bandwidth, and the desired tolerance to bit errors. Neither solution is appropriate for all cases. Section 8 defines parameters that may be used at session setup to choose between these approaches.3.7. Robustness against Packet Loss
The payload format supports several means, including forward error correction (FEC) and frame interleaving, to increase robustness against packet loss.3.7.1. Use of Forward Error Correction (FEC)
The simple scheme of repetition of previously sent data is one way of achieving FEC. Another possible scheme which is more bandwidth efficient is to use payload-external FEC, e.g., RFC 2733 [23], which generates extra packets containing repair data. The whole payload can also be sorted in sensitivity order to support external FEC schemes using UEP. There is also a work in progress on a generic version of such a scheme [22] that can be applied to AMR or AMR-WB payload transport. With AMR or AMR-WB, it is possible to use the multi-rate capability of the codec to send redundant copies of a frame using either the same mode or another mode, e.g., one with lower bandwidth. We describe such a scheme next.
This involves the simple retransmission of previously transmitted frame-blocks together with the current frame-block(s). This is done by using a sliding window to group the speech frame-blocks to send in each payload. Figure 1 below shows us an example. --+--------+--------+--------+--------+--------+--------+--------+-- | f(n-2) | f(n-1) | f(n) | f(n+1) | f(n+2) | f(n+3) | f(n+4) | --+--------+--------+--------+--------+--------+--------+--------+-- <---- p(n-1) ----> <----- p(n) -----> <---- p(n+1) ----> <---- p(n+2) ----> <---- p(n+3) ----> <---- p(n+4) ----> Figure 1: An example of redundant transmission In this example each frame-block is retransmitted one time in the following RTP payload packet. Here, f(n-2)..f(n+4) denotes a sequence of speech frame-blocks, and p(n-1)..p(n+4) a sequence of payload packets. The use of this approach does not require signaling at the session setup. However, a parameter for providing a maximum delay in transmitting any redundant frame is defined in Section 8. In other words, the speech sender can choose to use this scheme without consulting the receiver. This is because a packet containing redundant frames will not look different from a packet with only new frames. The receiver may receive multiple copies or versions (encoded with different modes) of a frame for a certain timestamp if no packet is lost. If multiple versions of the same speech frame are received, it is recommended that the mode with the highest rate be used by the speech decoder. This redundancy scheme provides the same functionality as the one described in RFC 2198, "RTP Payload for Redundant Audio Data" [27]. In most cases the mechanism in this payload format is more efficient and simpler than requiring both endpoints to support RFC 2198 in addition. There are two situations in which use of RFC 2198 is indicated: if the spread in time required between the primary and redundant encodings is larger than the duration of 5 frames, the bandwidth overhead of RFC 2198 will be lower; or, if a non-AMR codec is desired for the redundant encoding, the AMR payload format won't be able to carry it. The sender is responsible for selecting an appropriate amount of redundancy based on feedback about the channel, e.g., in RTCP
receiver reports. A sender should not base selection of FEC on the CMR, as this parameter most probably was set based on non-IP information, e.g., radio link performance measures. The sender is also responsible for avoiding congestion, which may be exacerbated by redundancy (see Section 6 for more details).3.7.2. Use of Frame Interleaving
To decrease protocol overhead, the payload design allows several speech frame-blocks to be encapsulated into a single RTP packet. One of the drawbacks of such an approach is that packet loss can cause loss of several consecutive speech frame-blocks, which usually causes clearly audible distortion in the reconstructed speech. Interleaving of frame-blocks can improve the speech quality in such cases by distributing the consecutive losses into a series of single frame- block losses. However, interleaving and bundling several frame- blocks per payload will also increase end-to-end delay and is therefore not appropriate for all types of applications. Streaming applications will most likely be able to exploit interleaving to improve speech quality in lossy transmission conditions. This payload design supports the use of frame interleaving as an option. For the encoder (speech sender) to use frame interleaving in its outbound RTP packets for a given session, the decoder (speech receiver) needs to indicate its support via out-of-band means (see Section 8).3.8. Bandwidth-Efficient or Octet-Aligned Mode
For a given session, the payload format can be either bandwidth efficient or octet aligned, depending on the mode of operation that is established for the session via out-of-band means. In the octet-aligned format, all the fields in a payload, including payload header, table of contents entries, and speech frames themselves, are individually aligned to octet boundaries to make implementations efficient. In the bandwidth-efficient format, only the full payload is octet aligned, so fewer padding bits are added. Note, octet alignment of a field or payload means that the last octet is padded with zeroes in the least significant bits to fill the octet. Also note that this padding is separate from padding indicated by the P bit in the RTP header. Between the two operation modes, only the octet-aligned mode has the capability to use the robust sorting, interleaving, and frame CRC to make the speech transport more robust to packet loss and bit errors.
3.9. AMR or AMR-WB Speech over IP Scenarios
The primary scenario for this payload format is IP end-to-end between two terminals, as shown in Figure 2. This payload format is expected to be useful for both conversational and streaming services. +----------+ +----------+ | | IP/UDP/RTP/AMR or | | | TERMINAL |<----------------------->| TERMINAL | | | IP/UDP/RTP/AMR-WB | | +----------+ +----------+ Figure 2: IP terminal to IP terminal scenario A conversational service puts requirements on the payload format. Low delay is one very important factor, i.e., few speech frame-blocks per payload packet. Low overhead is also required when the payload format traverses low bandwidth links, especially as the frequency of packets will be high. For low bandwidth links, it is also an advantage to support UED, which allows a link provider to reduce delay and packet loss, or to reduce the utilization of link resources. A streaming service has less strict real-time requirements and therefore can use a larger number of frame-blocks per packet than a conversational service. This reduces the overhead from IP, UDP, and RTP headers. However, including several frame-blocks per packet makes the transmission more vulnerable to packet loss, so interleaving may be used to reduce the effect that packet loss will have on speech quality. A streaming server handling a large number of clients also needs a payload format that requires as few resources as possible when doing packetization. The octet-aligned and interleaving modes require the least amount of resources, while CRC, robust sorting, and bandwidth-efficient modes have higher demands. Another scenario is when AMR or AMR-WB encoded speech is transmitted from a non-IP system (e.g., a GSM or 3GPP UMTS network) to an IP/UDP/RTP VoIP terminal, and/or vice versa, as depicted in Figure 3.
AMR or AMR-WB over I.366.{2,3} or +------+ +----------+ 3G Iu or | | IP/UDP/RTP/AMR or | | <------------->| GW |<---------------------->| TERMINAL | GSM Abis | | IP/UDP/RTP/AMR-WB | | etc. +------+ +----------+ | GSM/ | IP network 3GPP UMTS network | Figure 3: GW to VoIP terminal scenario In such a case, it is likely that the AMR or AMR-WB frame is packetized in a different way in the non-IP network and will need to be re-packetized into RTP at the gateway. Also, speech frames from the non-IP network may come with some UEP/UED information (e.g., a frame quality indicator) that will need to be preserved and forwarded on to the decoder along with the speech bits. This is specified in Section 4.3.2. AMR's capability to do fast mode switching is exploited in some non- IP networks to optimize speech quality. To preserve this functionality in scenarios including a gateway to an IP network, a codec mode request (CMR) field is needed. The gateway will be responsible for forwarding the CMR between the non-IP and IP parts in both directions. The IP terminal should follow the CMR forwarded by the gateway to optimize speech quality going to the non-IP decoder. The mode control algorithm in the gateway must accommodate the delay imposed by the IP network on the IP terminal's response to CMR. The IP terminal should not set the CMR (see Section 4.3.1), but the gateway can set the CMR value on frames going toward the encoder in the non-IP part to optimize speech quality from that encoder to the gateway. The gateway can alternatively set a lower CMR value, if desired, as one means to control congestion on the IP network. A third likely scenario is that IP/UDP/RTP is used as transport between two non-IP systems, i.e., IP is originated and terminated in gateways on both sides of the IP transport, as illustrated in Figure 4 below.
AMR or AMR-WB AMR or AMR-WB over over I.366.{2,3} or +------+ +------+ I.366.{2,3} or 3G Iu or | | IP/UDP/RTP/AMR or | | 3G Iu or <------------->| GW |<------------------->| GW |<-------------> GSM Abis | | IP/UDP/RTP/AMR-WB | | GSM Abis etc. +------+ +------+ etc. | | GSM/ | IP network | GSM/ 3GPP UMTS network | | 3GPP UMTS network Figure 4: GW to GW scenario This scenario requires the same mechanisms for preserving UED/UEP and CMR information as in the single gateway scenario. In addition, the CMR value may be set in packets received by the gateways on the IP network side. The gateway should forward to the non-IP side a CMR value that is the minimum of three values: - the CMR value it receives on the IP side; - the CMR value it calculates based on its reception quality on the non-IP side; and - a CMR value it may choose for congestion control of transmission on the IP side. The details of the control algorithm are left to the implementation.
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