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Versions: 00 01 02 03 04                                                
Network Working Group                            Lars Westberg, Ericsson
INTERNET-DRAFT                                Morgan Lindqvist, Ericsson
Expires: November 2000                                            Sweden
                                                            May 22, 2000

             Realtime Traffic over Cellular Access Networks

Status of this memo

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.

   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 cite them other than as _work in progress_.

   The list of current Internet-Drafts can be accessed at

   The list of Internet-Draft Shadow Directories can be accessed at

   This document is an individual submission to the IETF. Comments
   should be directed to the authors.


   The draft discusses problems with transport of realtime traffic over
   cellular access channels and their implications for protocol

Westberg, Lindqvist                                             [Page 1]

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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

Westberg, Lindqvist                                             [Page 2]

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   uses TCP and/or UDP. The cellular access is treated as a layer 2 (L2)
   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. Fewer users result in lower income, which in the end result in
   higher system cost.

   The cellular access network has wide area coverage and if the service
   requires more spectrum, the whole network capacity will be affected
   due to the mobility of the users. A poor spectrum usage for Voice
   over IP will cause much more cost for hardware in base stations than
   other more optimized services e.g. circuit switched voice.

Westberg, Lindqvist                                             [Page 3]

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   Another important factor is the cost of the spectrum itself. The
   spectrum is a limited resource and is licensed by the cellular
   operator. The cost for licensed RF-spectrum is estimated to be 30%
   [18] of the total operating cost and therefore represents a
   significant cost for the cellular operator.

   A typical circuit switched voice service requires a target BER of
   10e-3. If the BER requirement is changed from 10e-3 to 10e-6, this
   might result in a decreased spectrum efficiency of 25-50%, depending
   on the type of cellular system and the underlying radio conditions.
   This will increase system costs significantly. An efficient use of
   the spectrum resource is crucial.

2.2.  Lossy links - Design of link layer protocol based on radio

   If the radio link characteristics are not considered in the link
   layer design, the services will be more costly, or the performance in
   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.

   Delay in cellular systems is caused by several reasons. Forward Error
   Correction (FEC) and interleaving are used to increase the radio
   channel performance. A typical cellular radio channel gives, without
   any improvements, approximately 10% BER. With interleaving and FEC,
   the radio channel can provide the proper channel performance (as
   discussed above) for the service. But this introduces a fixed delay
   for the channel. In some cellular systems, the algorithmic channel
   RTT-delay is in the order of 80ms.

   Other sources of delay are DSP-processing, node internal delay and
   transmission. In all cellular system, the overall delay budget is

Westberg, Lindqvist                                             [Page 4]

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   optimized to achieve optimal performance for service quality and
   spectrum efficiency.

   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 for the case of radio channels optimized for real-time,
   100-200 ms if the speech coding is excluded. One may compare the RTT
   (Mobile->PSTN-GW->Mobile) for circuit switched speech, with 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.

   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

Westberg, Lindqvist                                             [Page 5]

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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
   covered 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
   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

Westberg, Lindqvist                                             [Page 6]

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   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 play-out 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 (real-time 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. However, a number of
   existing video compression standards (e.g. H.263 and MPEG4) have
   error-resilient modes that facilitate resynchronization of the
   decoder and improves performance on parts with errors in them.

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
   delivered to the application must be error-free. This is not a good
   solution for conversational applications, as was explained above.

Westberg, Lindqvist                                             [Page 7]

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   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.

Westberg, Lindqvist                                             [Page 8]

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5.  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).

   [14] Compressing IP/UDP/RTP Headers for Low-Speed Serial Links, IETF,
        RFC 2508.

Westberg, Lindqvist                                             [Page 9]

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   [15] CRTP over cellular radio links, IETF,
        draft-degermark-crtp-cellular-01.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.

   [18] Wireless Voice-over-IP and Implications for Third-Generation
        Network Design, Bell Labs Technical Journal, September 1998

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

This Internet-Draft expires November 22, 2000.

Westberg, Lindqvist                                            [Page 10]