Network Working Group Lars Westberg, Ericsson
INTERNET-DRAFT Morgan Lindqvist, Ericsson
Expires: December 1999 Sweden
June 21, 1999
Realtime Traffic over Cellular Access Networks
<draft-westberg-realtime-cellular-00.txt>
Status of this memo
This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026.
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This document is an individual submission to the IETF. Comments
should be directed to the authors.
Abstract
The draft discusses problems with transport of realtime traffic over
cellular access channels and their implications for protocol
enhancements.
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1. Realtime services over cellular access channels -
background and motivation
Emerging realtime services in the Internet, such as VoIP (Voice over
IP), impose new requirements on cellular access networks. Support for
these new services in cellular access networks may be provided in a
number of ways, ranging from interworking (e.g., terminating the IP
protocols in the fixed network and using other optimized protocols
over the cellular link) to transferring the IP packets end-to-end
over the cellular links. Transferring the IP packets end-to-end
allows the use of standard applications in the cellular terminal and
is therefore an important alternative.
Most of the work so far has been focused on transmission of best
effort traffic over wireless and not on the time critical
applications. End-to-end VoIP applications are possible to use in the
new generation of cellular networks, but the efficiency of radio
spectrum usage must be improved for such applications. Combining
spectrum efficiency, high quality speech and short delay calls for
new solutions.
The usual way to transport the IP packets in a radio network is to
use retransmissions over the radio link in order to obtain similar
characteristics as in the fixed network. This, however, will cause
long delays for speech, which in turn entails poor conversational
quality. Instead we need to solve the problems arising in the radio
network by enhance some parts of the protocol suite.
The scenario we are considering is one where two mobile stations
(MSs) are connected to a common fixed network through cellular links.
Mobile Base Base Mobile
Station Station Station Station
! ~ ~ ~ ~ ~ ~ ~ Y Y ~ ~ ~ ~ ~ ~ ~ !
! ! ! !
!----! ! ! !----!
! ! ! ! ! !
! MS ! ! ! ! MS !
!----! !++++++++! !----!
Fixed Network
The mobile stations contain a Voice-Over-IP application and a full IP
stack. The application generates audio, video and application-
specific session signaling, e.g., SIP/H.323. The audio/video is
transported over RTP/UDP/IP, while the application-specific signaling
uses TCP and/or UDP. The cellular access is treated as a layer 2 (L2)
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network with functionality for optimizing the performance on the
cellular link.
In this document, we summarize some of the requirements on the layers
that must be met in order to achieve good speech quality and spectrum
efficiency when transferring IP packets transparently over the
cellular access. It should be noted that due to the spectrum cost of
the transparent solution, alternative solutions such as interworking
also deserve to be considered. Such solutions, however, are not
discussed in this draft.
It should also be pointed out that some of the problems with realtime
packets over cellular access might only be solvable with "wireless
aware" terminals, meaning that not only the link layers, but also the
IP stack must be "wireless aware". However, all terminals and
applications will not be "wireless aware". Interworking between the
two classes of terminals/applications can be solved by gateways in
the fixed network.
2. Cellular access performance and system cost
The cellular radio access puts tough requirements on end-to-end
packet transmission. Packet transmission over the cellular access is
typically constrained by two factors:
- The high cost of cellular access links. Cellular bandwidth with
high quality imposes high system cost.
- The lossy link behavior. The radio network generates a high BER.
If retransmission over the radio link is not used, the BER may
be in the order of 10e-3.
2.1. System Cost - Selection of BER for the radio link
In wireless systems there is a close relationship between the BER and
the SNR (signal-to-noise ratio) of the radio channel. Furthermore,
the required SNR (corresponding to a selected BER requirement) can be
directly related to the system capacity, i.e. the number of users per
cell. Less users result in lower income, which in the end result in
higher system cost. Requiring a BER of 10e-6 instead of 10e-3 might
result in an increase in system cost of between 30% and 100%,
depending on the type of cellular system and the underlying radio
conditions.
2.2. Lossy links - Design of link layer protocol based on radio
requirements
If the radio link characteristics are not considered in the link
layer design, the services will be more costly, or the performance in
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terms of speech quality will be poor. The design of current protocols
is based on the transmission characteristics of fixed networks, so
these are not well suited to the radio requirements. A good trade-off
between the requirements of transmission (BER and packet loss) and
the design of the protocols is crucial.
The quality in a radio system (expressed for instance as the BER)
typically changes from one 20 ms radio frame to another due to fading
in the radio channel. For example, in a cellular system where the
target BER has been set to 10e-3, the BER might vary, roughly
speaking, from 10e-4 to 10e-2. However, the average BER will equal
the target BER. Thus, for the system capacity to remain unaffected,
the link layer protocols and speech coders must tolerate that the BER
exceeds the BER target for limited periods of time.
Another important aspect is the long round-trip time (RTT). Even if
we use channels without retransmission over the radio link, the
unidirectional delay might be important to consider for some link
layer functions, such as header compression. It is difficult to state
a generally valid value for the RTT. Some RTT figures (e.g. 200 ms)
are mentioned in [16], but the delay might be shorter in the case of
real-time channels (100-200 ms). To this one may compare the RTT for
circuit switched speech, the "long-term objective value" which is
stated to be less than 180 ms for GSM-FR (GSM full-rate speech codec)
in [17].
3. Transport of realtime IP flows over cellular
We summarize the problems of transporting realtime packets over
cellular links and the implications of these problems for protocol
enhancements in wireless transmission.
3.1. Layer 2 enhancements for realtime traffic
To efficiently transfer audio and video streams over the radio
channels, these flows should be identified and de-multiplexed.
Identification of realtime flows could be carried out by heuristic
rules, as proposed in [12]. One of the problems is that the radio
channels still need to be adapted to the characteristics of the
compressed information. The BER assignment might be different for
audio and video. We might also differentiate the redundancy coding of
the compressed payload, something that requires a detailed knowledge
of the payload.
Therefore, for RTP flows that have dynamically assigned payload type
indicator (PTI) values [13], the identification of codec type is
important in order to allow simple layer 2 identification of the
compressed payload type.
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Apart from the problems of transporting the payload, we also have to
perform optimization of the protocol headers [14] for real-time
traffic. One of the major problems here is that the compression
algorithm must work well in the radio environment with its link
delay, and also be resistant to bit errors. The current header
compression scheme [14] is sensitive to bit errors, as is shown in
[15].
3.2. Transport of audio over cellular
3.2.1. Properties of speech codecs designed for cellular networks
A cellular access link is very lossy and expensive compared to fixed
lines. Furthermore telephony is a real-time service where
retransmissions should be avoided. Thus the speech decoder should
receive not only error-free speech frames but also frames with errors
[1]. If all erroneous speech frames are dropped, the frame error rate
(FER) will be high. With a high FER, it is not possible to produce
speech with an acceptable quality. In cellular telephony systems,
this problem is overcome by delivering all frames to the speech
decoder, regardless of bit errors.
The sensitivity to errors varies widely between different bits in a
frame of encoded speech. High error sensitivity means that an error
in that bit results in a severe degradation in speech quality. In
most cellular speech codecs for 2nd generation mobile systems (GSM,
TDMA or PDC), the bits are divided into three classes: 1a, 1b and 2.
Class 1a (the most sensitive bits) and 1b (medium sensitive bits) are
protected by convolutional coding. Class 1a bits are in addition
protected by a CRC. Class 2 bits (the least sensitive bits) are not
protected at all. This scheme results in a reduced FER, since a frame
is considered erroneous only if there are errors in the class 1a bits
(which on average amount to one third of the total number of bits).
On the other hand, the scheme also leaves undetected residual errors
in class 1b and class 2. However, it is better, from a speech quality
point of view, to allow some errors in these bits than to discard the
whole frame as soon as bit errors occur, and let the ECU (see 3.2.2
below) reconstruct the frame [2][3][4].
3.2.2. Error Concealment Unit (ECU)
If the CRC for the class 1a bits is corrupt, there are severe errors
in the speech frame which probably would give rise to annoying
distortions. The frame is therefore discarded, and instead the speech
decoder generates artificial speech that closely resembles that of
the previous frames. In this way the decoder attempts to mask the
distortion. The component carrying out this task is called the error
concealment unit (ECU). The ECU reconstructs the frame based on the
corrupt version of it that was received as well as the last good
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frame. Using the bad frame itself increases the speech quality if the
number of damaged bits in the frame is not too high.
In most speech coders with a 20 ms frame size, the ECU is a state
machine with 6 states. When several consecutive lost/bad frames are
encountered, the ECU proceeds to the next state for each new frame
(until it reaches state 6). The amplitude of the generated speech is
gradually reduced in the states, and in state 6, the speech is
completely muted.
If the decoder receives a good frame when the ECU is in state 1-5, it
immediately switches back to normal decoding mode. If the ECU is in
state 6 when a good frame is received, the decoder awaits the next
frame. If this frame is also good, the decoder switches back to
normal decoding mode, otherwise it remains in state 6. This feature
is added to eliminate sound spikes when recovering from a long
sequence of lost/bad frames. It also mitigates the effects of
residual bit errors in the class 1a bits.
3.2.3. Adaptive buffer management
In order to minimize the end-to-end delay, an adaptive buffer manager
(ABM) is useful, another term is adaptive playout buffer. The
function of the adaptive buffer manager is to change the buffer size
in order to allow as many packets as possible to reach the speech
decoder in time, while keeping the buffering delay to a minimum.
To achieve good performance, the ABM should treat the packets with
bit errors in the payload as normal packets, not as late packets.
Otherwise, the buffer size might be larger than necessary.
3.3. Transport of Video
The transport of compressed video, intended for conversational
services (i.e., videophone) over cellular links, entails some unique
requirements. One is that, delay must be kept under strict control.
Cellular links also have other error characteristics than fixed
networks, something that may cause problems.
One way to fulfil the requirements (realtime and limited delay) is to
allow bit errors in the payload in the same way as for speech. Errors
in the payload are of course not permissible for all type of packets,
and not even for all video streams, but a number of existing video
compression standards do accept errors in the compressed bit stream,
notably H.263 and MPEG4.
3.3.1. Conversational video in a wireless environment
Wireless channels have high bit error rates. These high bit error
rates will result in requirements on retransmission, if all packets
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delivered to the application must be error-free. This is not a good
solution for conversational applications, as was explained above.
There are, at least, two possible ways to implement a conversational
service over channels with high bit error rate. One is to use strong
forward error correction; another is to have an error-robust method
of video compression (usually called error-resilient source coding).
Many experiments (ITU, MPEG, ARIB, 3GPP) have shown that for a
wireless channel, error-resilient source coding [5] outperforms
methods using forward error correction codes.
4. Conclusions
Spectrum efficient transmission of audio and video packets is
extremely important in cellular access. Spectrum constitutes a
significant cost for the operator and must be considered in the
development of end-user services. To facilitate effective transport
of voice and video in a cellular system, some improvements of the IP
protocol suite are needed. Some of the changes are related to the
link layer and some to the behavior of RTP (real-time protocol).
The following improvements are identified:
- Simple identification of codec type in the link layer. The
knowledge of codec type enables enhancement of the performance
over the wireless link.
- BER-resistant header compression algorithm for RTP/UDP/IP. The
header compression algorithm also has to work well in an
environment with long round-trip delays.
- No dropped packets due to bit errors in the payload. The speech
decoder and the buffer manager perform better if they can access
all packets, also those that contain bit errors.
- Use of CRC for the most sensitive bits in the payload in order
to detect bit errors. This improves the performance of the
speech decoder.
6. Authors' addresses
Lars Westberg
Ericsson Research
E-mail: rtiow@era-t.ericsson.se
Morgan Lindquist
Ericsson Research
E-mail: morgan.lindqvist@era.ericsson.se
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7. References
[1] European digital cellular telecommunication system (Phase 2):
Radio transmission and reception (GSM 05.05), European
Telecommunications Standards Institute, October 1993.
[2] European digital cellular telecommunication system (Phase 2):
Channel coding (GSM 05.03), European Telecommunications
Standards Institute, October 1993.
[3] TIA/EIA/IS-641, Interim Standard, TDMA Cellular/PCS radio
interface - Enhanced Full-Rate Speech Codec, May 1996.
[4] Association of Radio Industry and Business, RCR STD-27F, 1997.
[5] Signal Processing, Image Communication, Special Issue on Error
Resilience, Volume 14, Nos. 6-8, May 1999, page 443-676, ISSN
0923-5965, Guest Editors: J.C Brailean, T. Sikora and T. Miki.
[6] Video coding for low bit rate communication, Recommendation
H.263 (02/98), International Telecommunication Union.
[7] Multiplexing protocol for low bit rate multimedia communication,
Recommendation H.223 (03/96) with later annexes (A,B,C 02/98),
International Telecommunication Union.
[8] Information Technology -- Very low bitrate audio-visual coding -
Part 2: Visual, ISO/IEC 14496-2 ("MPEG4").
[9] Association of Radio Industry and Business, Test results of
video multimedia codec simulation, ICWG 16-4, July 17, 1998.
[10] Association of Radio Industry and Business, Report of ARIB IMT-
2000 Video Multimedia Codec Simulation Test, ICW-VMG35-, March
18, 1999.
[11] 3rd Generation Partnership Project (3GPP), TSG-SA Coding Working
Group, "QoS for Speech and Multimedia Codec Quantitative
performance evaluation of H.324 Annex C over 3G", TR 26.116
(working document).
[12] Heuristics for utilizing ISSL Mechanisms for A/V Streams over
Low Bandwidth Links in the absence of Announcement Protocols,
IETF, draft-putzolu-heuristic-00.txt (work in progress).
[13] RTP Profile for Audio and Video Conferences with Minimal
Control, IETF, ietf-avt-profile-new-05.txt (work in progress).
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[14] Compressing IP/UDP/RTP Headers for Low-Speed Serial Links, IETF,
RFC 2508.
[15] CRTP over cellular radio links, IETF,
draft-degermark-crtp-cellular-00.txt (work in progress).
[16] Long Thin Networks, IETF, draft-montenegro-pilc-ltn-02.txt
(work in progress).
[17] European digital cellular telecommunication system (phase 1):
Technical Performance Objectives, GSM 03.05, version 3.2.0.
This Internet-Draft expires in December 1999.
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