RFC 3267 PDF

Заголовок UDP-Lite должен всегда покрываться контрольной суммой. Это означает, что отличные от 0 значения поля Checksum Coverage должны быть не меньше Безотносительно к значению Checksum Coverage поле Checksum должно учитывать псевдозаголовок, основанный на заголовке IP см. Поле Checksum содержит битовое дополнение до 1 суммы дополнений для информации псевдозаголовка, собранной из заголовка IP, числа октетов, заданного полем Checksum Coverage начиная с первого октета в заголовке UDP-Lite , возможно дополненного октетом нулей в конце для выравнивания по битовой границе [RFC].

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Sjoberg Request for Comments: M. Lakaniemi Nokia Q. 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.

All Rights Reserved. Sjoberg, et. Conventions and Acronyms Multi-rate Encoding and Mode Adaptation Voice Activity Detection and Discontinuous Transmission Support for Multi-Channel Session Unequal Bit-error Detection and Protection Robustness against Packet Loss Use of Frame Interleaving Bandwidth Efficient or Octet-aligned Mode RTP Header Usage Payload Structure Bandwidth-Efficient Mode The Payload Header The Payload Table of Contents Speech Data Algorithm for Forming the Payload Octet-aligned Mode Methods for Forming the Payload Payload Examples Implementation Considerations Single Channel Header Multi-channel Header Speech Frames Congestion Control Security Considerations Decoding Validation Payload Format Parameters IANA Considerations Full Copyright Statement 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. 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.

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.

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. The AMR codec is a multi-mode codec that supports 8 narrow band speech encoding modes with bit rates between 4. The sampling frequency used in AMR is Hz and the speech encoding is performed on 20 ms speech frames. Therefore, each encoded AMR speech frame represents samples of the original speech. Particularly, the 6.

AMR-WB supports 9 wide band speech coding modes with respective bit rates ranging from 6. Multi-rate Encoding and Mode Adaptation The multi-rate encoding i. To perform mode adaptation, the decoder speech receiver needs to signal the encoder speech sender the new mode it prefers.

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 Sjoberg, et. 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.

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 MIME parameters that may be used to signal these mode adaptation controls at session setup. 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.

Support for Multi-Channel Session Both the RTP payload format and the storage format defined in this document support multi-channel audio content e. 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 Sjoberg, et. This property has been exploited in cellular systems to achieve better voice quality by using unequal error protection and detection UEP and UED mechanisms. A frame is only declared damaged if there are bit errors found in the most sensitive bits, i.

On the other hand, it is acceptable to have some bit errors in the other bits, i. 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. Today there exist some link layers that do not discard packets with bit errors, e. With the Internet traffic pattern shifting towards a more multimedia-centric one, more link layers of such nature may emerge in the future.

There are at least two basic approaches for carrying AMR and AMR-WB traffic over bit error tolerant IP networks: 1 Utilizing a partial checksum to cover headers and the most important speech bits of the payload. It is recommended that at least all class A bits are covered by the checksum. Note, it is still important that the network designer pay 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. Approach 1 is a bit efficient, flexible and simple way, 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 frames in a payload there is the possibility that a single bit error in protected bits will cause all the frames to be discarded.

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 gives an improvement in speech quality when transporting multiple frames over links subject to bit errors. The choice between the above two approaches must be made based on the available bandwidth, and desired tolerance to bit errors.

Neither solution is appropriate to all cases. Section 8 defines parameters that may be used at session setup to select between these approaches.

Robustness against Packet Loss The payload format supports several means, including forward error correction FEC and frame interleaving, to increase robustness against packet loss. Another possible scheme which is more bandwidth efficient is to use payload external FEC, e.

We describe such a scheme next. 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. In this example each frame-block is retransmitted one time in the following RTP payload packet. Here, f n The use of this approach does not require signaling at the session setup. 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. In most cases the mechanism in this payload format is more efficient and simpler than requiring both endpoints to support RFC in addition.

The sender is responsible for selecting an appropriate amount of redundancy based on feedback about the channel, e. The sender is also responsible for avoiding congestion, which may be exacerbated by redundancy see Section 6 for more details. Use of Frame Interleaving To decrease protocol overhead, the payload design allows several speech frame-blocks be encapsulated into a single RTP packet.

One of the drawbacks of such an approach is that in case of packet loss this means loss of several consecutive speech frame-blocks, which usually causes clearly audible distortion in the reconstructed speech.


RFC 3268 PDF

Handley Request for Comments: V. 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. All Rights Reserved.


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