The Audio Gateway is one of the VE-MASE's media gateways and provides for real-time audio conferencing between peers in a low bandwidth wireless access ...
Voice/Data Integration in Wireless Communication Networks Michael Wallbaum, Jens Meggers Department of Computer Science 4, RWTH Aachen, Germany Email: {wallbaum | meggers}@i4.informatik.rwth-aachen.de
Abstract: Currently the integration of voice and data in mobile communication networks is still in its infancy. Restrictions to multimedia communication are placed by changing quality of service and the use of separate voice and data bearer services. This paper describes how the middleware architecture developed in the ACTS project MOVE provides for voice/data integrated communication even over a single data bearer service.
1. Introduction The Internet offers a wide range of services ranging from data services such as the World Wide Web (WWW), electronic mail and file transfer to interactive multimedia applications. The packet-switched nature of the Internet Protocol (IP) facilitates the simultaneous use of different services across different networks. On the contrary, today’s usage of wide area mobile networks is mainly limited to circuitswitched voice connections, fax and short message services. Although undoubtedly useful, these services do not provide adequate support for multimedia applications because they do not well synchronise the different media, especially voice and data. Additional problems are caused by the currently limited bandwidth of around 10 kbit/s and the circuit-switched nature of the bearer services which do not provide for efficient network utilisation. Though future mobile networks will offer real-time packet data bearer services and higher bandwidths, mobile communication will always suffer from dynamically changing quality of service (QoS) which requires the adaptation of the different media streams. In preparation for future 3 rd generation mobile networks the ACTS project MOVE [1] currently designs and develops a middleware architecture
called Voice-Enabled Mobile Application Support Environment (VE-MASE). The VE-MASE enhances the middleware architecture, which was developed in the ACTS project OnTheMove [2], by providing support for interactive real-time multimedia applications and integrated voice and data services. The aim is to enable a completely new class of interactive multimedia services targeted at, but not limited to, mobile devices that are equipped with the VE-MASE. This paper discusses the VEMASE components required for audio communication and for the integration of voice and data services. Section 2 of this paper will briefly introduce the VE-MASE middleware architecture and describe its components. Section 3 will then discuss the so-called Audio Gateway as an example of one of the media adaptation gateways which are part of the VE-MASE. The problem of co-ordinating the adaptation the different media streams is then discussed in Section 4. This section also describes the operation of the System Adaptability Manager which is responsible for quality of service trading. A conclusion and outlook is presented in Section 5.
2. MOVE Middleware Architecture The VE-MASE enhances the Mobile Application Support Environment (MASE), that was developed in the course of the OnTheMove project. The MASE provides for seamless integration of different bearers, carriers and terminal types with a focus on ”static” multimedia services such as the delivery of textual information and images. It was assumed that the mobile user only generates and sends a limited amount of data.
GSM ISDN / PSTN / GSM
Wireless Cellular Network LAN
VE-MASE Mobility Gateway
Data network (e.g., Internet)
VE-MASE equipped Service Provider
UMTS
Figure 1: Overview Of The VE-MASE Architecture The VE-MASE refines the existing MASE components and adds new components to create a true multimedia infrastructure enabling bidirectional, real-time services. Figure 1 illustrates the MOVE approach: Mobile devices connect to the Mobility Gateway, which acts as a mediator between mobile and fixed network. The gateway performs network- and applicationspecific adaptation and media conversion. Notably, adaptation and conversion can be done across boundaries of different networks, thus concealing the heterogeneity of underlying bearer services and mobility-related functions. The VE-MASE is distributed and partially replicated over mobile devices, Mobility Gateway and service provider. The main components of the VE-MASE are the System Adaptability Manager (SAM), the Audio Gateway, the Multimedia Conversion Proxy (MMC-Proxy) and the Scheduler. The SAM collects events and measurements from the relevant VE-MASE components to determine the currently available QoS. It then instructs the media gateways of how to adapt the streams to meet the current conditions. The Audio Gateway and the MMC-Proxy adapt audio and data streams according to the SAM. The Audio Gateway will be described in the next section. Finally, the Scheduler ensures that real-time streams are not delayed by non real-time data. Incoming packets
are classified according to their service class associated with a priority queue that has a specific delay and jitter bound property. A detailed description of the VE-MASE components is given in [3][4].
3. The Audio Gateway The Audio Gateway is one of the VE-MASE’s media gateways and provides for real-time audio conferencing between peers in a low bandwidth wireless access network and peers located in the fixed network environment. It is based upon UCL’s Robust Audio Tool RAT [5] and thus realises an RTP transcoder [6][7]. As a mediator between the participants in the fixed and wireless networks the gateway intercepts the data streams on the application level to perform bandwidth adaptation and to increase the robustness of the audio streams to packet loss. The Audio Gateway’s main feature is its capability to change the encoding of an incoming audio stream. This capability can be deployed in the following scenarios: 1. The terminals used in an audio conference do not share a common audio codec that can be employed under the current network conditions.
40000
35000
Bandwidth [bit/s]
30000
25000
20000
15000
10000
5000
0 0
250
500
750
1000
1250
1500
1750
2000
2250
t [sec/2] Transcoder Input
Transcoder Output
Figure 2: Bandwidth Reduction On One Link Of A VoIP Session. 2. The Audio Gateway can perform bandwidth adaptation when the codec agreed upon cannot be used due to temporarily reduced bandwidth on the wireless link, e.g. caused by the user’s movement or shadow and multipath fading. Conversely, a better codec with a higher bitrate could be employed to improve the quality of audio playback, if the bandwidth increases. In some cases the change in the quality of the wireless link can be expected to stay persistent over a longer period of time, e.g. due to a handover to a different access network. In such situations the adaptive VE-MASE voice over IP (VoIP) client can be instructed to deploy a different codec more suitable for the current network conditions. This is more efficient in terms of bandwidth consumed on the fixed network, computing power on the Audio Gateway and transcoding delays.
payload was measured at the Audio Gateway on the input and output of a link from a fixed to a wireless network client. The fixed network client emits an audio stream encoded in DVI [8][9] with no silence suppression, which consumes around 32 kbit/s. At t=1200 the fixed network sender turned on silence suppression reducing the consumed bandwidth to an average of 22.3 kbit/s. The gateway transcodes the incoming audio frames using a proprietary vocoder, which yields a bandwidth of 3.2 kbit/s. Furthermore silence suppression is performed for the whole duration of the conference resulting in an average bandwidth of 1.9 kbit/s. Thus, if IP/UDP/RTP-header compression is used [10] then VoIP sessions can even be conducted over a GSM data bearer service. Note that an additional advantage of the Audio Gateway is that it can also perform forward error correction by redundant encoding of audio data [11] to increase the robustness of the audio client to packet loss.
Besides changing the encoding of the audio data the Audio Gateway can also adapt bandwidth by suppressing silence in the outgoing audio streams and changing the amount of audio data per packet to reduce header overhead. Figure 2 demonstrates the effect of the gateway’s adaptation capabilities. The net bandwidth of the audio
As stated before, the Audio Gateway is only one example for the media gateways supported by the VE-MASE. Other gateways which were not discussed here perform bandwidth adaptation for video [12] and HTTP-streams, and reduce the size of email attachments when using the IMAP protocol.
4. Voice/Data Integration The media gateways are essential components of the VE-MASE but they only provide for the adaptation of the data. The question is how the streams should be adapted and what the trade-off between modifications to different streams is. The SAM was introduced in the OnTheMove project to perform QoS trading under the assumption that only one data stream at a time is transmitted over the wireless link. Other research groups have worked under similar assumptions [13]. This QoS trading policy is insufficient, since the aim of the MOVE project is to provide for the integration of real-time and non-real-time streams being transmitted simultaneously. The enhanced SAM takes into account how simultaneous voice and data streams interact and affect each other. It collects events and measurements from other VE-MASE components, performs real-time analysis of QoS parameters for realand non-real-time data, and instructs the gateways of new adaptation settings. Best
High
MOS 3.8–4.2 Quality 4.2–5.0 Mouthto-ear- 0–150 ms 150–250 ms delay Call setup time
0–1 sec
1–3 sec
Medium
Low
Best Effort
2.9–3.8
2.0–2.9
1.0–2.0
250– 350 ms
350– 500 ms
≥ 500 ms
3–5 sec
5–10sec
≥ 10 sec
Table 1: Voice Quality Classification After collecting the events and measurements from the other components it classifies the current QoS of each medium and compares it to the user’s preferences as stated in the profile. These preferences are expressed in terms of the lowest quality accepted by the user for each medium. Due to the variety of QoS parameters the user’s choice is restricted to several predefined QoS classes. Table 1 shows the QoS classes defined for VoIP sessions. The table is derived from a classification elaborated in [14]. The additional QoS class “Low” takes into account that the quality of voice transmission via IP currently is
Collector
Link Trader
'DWD 7UDGHU
9RLFH 7UDGHU
User Profile
Notifier
Figure 3: SAM Architecture often considerably below common standards of conventional telephony If the current QoS of a media stream does not comply with the user’s specifications, then the SAM’s link trader requests new adaptation settings from the media traders. These QoS traders will try to determine new adaptation settings which reduce the bandwidth requirements, but still corresponds with the chosen QoS class. If the bandwidth of one stream can be reduced, such that all media streams can be maintained, then the media gateways are instructed of the new adaptation settings. If it is not feasible to maintain the state of all streams according to the desired quality, then one or more streams will be terminated. For example, in a combined VoIP and collaborative web browsing session, the latter could be shut down, if this measure can maintain the VoIP session. The SAM architecture as depicted in Figure 3 is described in greater detail in [3].
5. Conclusion and Future Work In this paper we discuss the approach of the MOVE project to integrate voice and data in mobile communication networks. The proposed middleware architecture called VE-MASE has been implemented to a great extent in the form of a demonstrator. A sample call-centre application using the VE-MASE has also been developed.
This application enables a mobile user to browse a hotel search service and call the receptionist to make a reservation by pressing a so-called “Call Button” on the webpage. Future work will concentrate on evaluating the middleware, conducting measurements to determine the transcoding delays involved and to test the system with different wireless access networks.
[10]
[11]
[12]
[13]
References [1] [2]
[3]
[4]
[5]
[6]
[7]
[8]
[9]
ACTS MOVE Homepage: http://move.rwthaachen.de. B. Kreller, A. Park, J. Meggers, G. Forsgren, E. Kovacs, M. Rosinus: „UMTS: A Middleware Architecture and mobile API Approach“, IEEE Personal Communications Magazine, April 1998. H. Decker, M. Krautgärtner, C. Ong, M. Wallbaum: “Quality of Service Management in an Integrated Mobile Voice/Data-Enabled Service Architecture”, Proceedings of the 4th ACTS Mobile Communications Summit, Sorrento, Italy, June 1999. M. Wallbaum, D. Carrega, M. Krautgärtner, H. Decker: “A Mobile Middleware Component Providing Voice Over IP Services To Mobile Users”, Proceedings of ECMAST'99, Lecture Notes in Computer Science Vol. 1629: Leopold, H.; Garcia, N., (Eds.) Multimedia Applications, Services and Techniques - ECMAST'99, Heidelberg, Germany, May 1999. C. Perkins, V. Hardman, I. Kouvelas, A. Sasse: “Multicast audio: The next generation”, In International Networking Conference (INET), Kuala Lumpur, Malaysia, June 1997. H. Schulzrinne et. Al: “RTP: A Transport Protocol for Real-Time Applications“, IETF Request For Comments 1889, January, 1996. H. Schulzrinne, “RTP Profile for Audio and Video Conferences with Minimal Control”, IETF Request For Comments 1890, January 1996. R. V. Cox: „Three Speech Coders from the ITU Cover a Range of Applications“, IEEE Communications Magazine, November 1997. R.V.Cox, P.Kroon: “Low bit-rate speech coders for multimedia communication”, IEEE Communication Magazine, pp.34-41, December 1996.
[14]
S. Casner, V. Jacobson: “Compressing IP/UDP/RTP Headers for Low-Speed Serial Links”, RFC 2508, February 1999. C. Perkins, et al: „RTP Payload for Redundant Audio Data“, IETF Request For Comments 2198, September 1997. J. Meggers, T. Strang, A. Park: „A Video Gateway to Support Video Streaming to Mobile Clients“, ACTS Mobile Communication Summit, Aalborg, October 1997. S. N. Bhatti, G. Knight: “Enabling QoS adaptation decisions for Internet applications”, Journal of Computer Networks, vol. 31, no. 7, pp 669-692, March 1999. ETSI TIPHON, “Telecommunications and Internet Protocol Harmonization Over Networks (TIPHON); General aspects of Quality of Service (QoS)“, TR 101 329 V2.1.1 (199906), June 1999.
