Enhanced Voice over IP Support in GPRS and EGPRS

Transcription

1 Enhanced Voice over IP Support in GPRS and EGPRS Andreas Schieder*, Tobias Ley Ericsson Research, Ericsson Eurolab Deutschland Ericsson Allee 1, Herzogenrath, Germany * Abstract- In this paper we discuss the applicability of GPRS and EGPRS for the transmission of voice packets in an end-toend packet switched scenario. We assume typical Voice over IP (VoIP) applications spanning between a fixed and a mobile terminal. As the transmission delay of the voice packets has most impact on the perceived Quality of Service (QoS), it has been put into focus of our investigations. The studies have been aided by an end-to-end simulation environment modeling the protocol stack applied in (E)GPRS. The first assessments performed were dedicated to obtain performance figures of standard (E)GPRS systems carrying VoIP data. In a second step, the mechanisms and procedures having most relevance to the transmission delay in (E)GPRS are identified and improvements are proposed. These improvements start from smaller modifications of the standard system and end at modifications of the multiplexing schemes applied on the radio interface. As we will see, the typical transmission delay of (E)GPRS is thus stepwise improved at the expense of increased modifications of the system. I. INTRODUCTION The current trend towards a unique transmission technology for all conceivable telecommunication services, which can be observed for fixed networks, led to solutions assuming IP as ubiquitous transmission technology applied in fixed networks. This trend mainly stems from convergence forces merging traditional telecommunication and datacommunication into a single communication world making the development and deployment of new services more convenient and economical. Still the agreement on a uniform protocol is only the first step. The main challenge in achieving a true convergence is ensuring the QoS required by the services, especially realtime services, in such an all-ip network [1]. Those networks are typically suited to convey data services such as WWW browsing, but need improvements and additions in order to make realtime applications such as traditional telephony possible. Improvements are definitely required for wired networks, but more challenging is the task to allow realtime applications over packet switched mobile networks, such as GPRS or the EDGE enhanced version of GPRS. In this paper we present performance figures, which have been obtained by extensive simulations for the application of VoIP over GPRS and EGPRS. A number of improvements, which take into account the limitations arising from the required conformance to the standardized elements of (E)GPRS, are introduced and the quantitative effects on typical performance parameters are presented. Main focus is put on the end-to-end delay the voice-packets experience. II. GPRS / EGPRS GPRS relies on the same modulation schemes as the circuit switched services in GSM do. The TDMA scheme applied in GSM is also kept, only coding schemes and access control for the timeslots in the TDMA structure have been changed to improve the support of packet switched transmission. EDGE (Enhanced Data rates for GSM Evolution) introduces new modulation schemes to increase the data rate per timeslot. Additionally, two new concepts, Link Adaptation (LA) and Incremental Redundancy (IR), are introduced to improve the efficient usage of the scarce radio resources, without loosing the possibility to operate EDGE under poor radio conditions. Both schemes are thus used to control the amount of redundancy added to the payload and the robustness against bit-errors resulting into optimized usage of the radio resources in dependence of the channel conditions. Both, GPRS and the EDGE enhanced EGPRS, incorporate identical multiplexing mechanisms applied on the radio interface. Due to this the following performance discussions are applicable for both systems. We will therefore concentrate on EGPRS, which realizes nine different channel coding schemes offering higher channel bit-rates and lower error protection with increasing index of the Modulation and Coding Scheme (MCS). The GPRS coding schemes and the GMSK modulation used in GPRS are incorporated by MCS1-4 of EGPRS (see [2], [7] and [8] for more information on EDGE and EGPRS). The scheduling mechanisms applied in both systems, from now on denoted (E)GPRS, allow the dynamic assignment of radio capacity to active sessions on Packet Data Channels (PDCH), which are determined by frequency and timeslot position in the GSM TDMA structure. The multiplexing for both, the downlink and uplink direction, is controlled by the network, i.e. the BSS. In uplink direction the (E)GPRS MSs can be multiplexed dynamically on the PDCH through identifying the MS by USFs (Uplink State Flags) [8]. Before being allowed to transmit data, the MS has to establish a TBF (Temporary Block Flow) requiring the exchange of signaling messages. That way, a physical connection between the MS and the network is established, which provides unidirectional transfer means for LLC (Logical Link Control) PDUs. If the TBF is established, the USF is scheduled on the downlink to handle the resource management of the MS. When receiving a USF, the MS is allowed to send on the same TS number in uplink direction on which it has received the USF. The establishment of a TBF takes about 1 to 15 ms for the uplink and 6 to 1 ms for the downlink direction in the ideal case, i.e. if the request message could be sent successfully on the random access channel. According to the

2 standard the TBF should be kept alive as long as the sending entity has data in its queue. III. QOS In order to have a basis for discussion of the delay values, presented in the following sections, one needs to know the delay bounds, which have been identified for the transmission of speech information. According to [3] the 3GPP has defined 4 ms to be the upper limit for the transmission delay of data stemming from audio applications. Nevertheless, one should aim at 15 ms end-to-end delay if possible, as the perceived service quality starts dropping, when this threshold is exceeded (see [3]). IV. SIMULATIONS The simulation environment used in our studies comprises all relevant (E)GPRS nodes including detailed models for the relevant protocols. The simulator models one cell with 8 PDCH for (E)GPRS on a single frequency (see [4] for a more detailed description of the system). The used audio source models the behavior of speakers in a conversation. The talkspurt length and idle time is exponentially distributed with a mean length of 135 ms and 165 ms respectively. It is assumed that the speech coder G is applied within an H.323 session. Each H.323 call is established once at the beginning of the simulation and kept alive for the whole simulation time. The voice packets generated by the speech coder are encapsulated in RTP, UDP and IP before being passed to the (E)GPRS network. Due to the significantly high overhead and the comparatively small speech frame size, it is necessary to demand header compression in (E)GPRS. In the following simulations ideal header compression is assumed for the combined header of RTP, UDP and IP, reducing the header to 4 bytes. The simulations have been started with analyses of a standard (E)GPRS system. The main outcome of these investigations was the discovery that the standardized TBF handling [7] does not meet the requirements of realtime applications. The procedures and rules defined for establishing, maintaining and releasing such TBFs increased the end-to-end delay in such a way that the thresholds defined in section 2 were exceeded quickly. In cases where EGPRS was able to offer a higher channel data rate than the actual data rate generated by the VoIP application many TBF releases and following re-establishments were observed. The reason is that the design of (E)GPRS has been based on the assumptions that larger application data packets should be transmitted in a single TBF. Those packets are typically generated by non-realtime applications such as WWW browsing. In case of a continuous stream of small application packets, as it is the case for VoIP, this approach can result in a stop-and-go transmission characterized by frequent TBF releases and re-establishments. Due to this the delay requirement of 15 ms can not be met even in optimal cases, i.e. low network load and a good radio channel. In the following sections we will focus on the assessment of the uplink performance, as it usually exhibits a lower QoS 95-percentile of end-to-end delay [ms] than the downlink due to the fundamental design characteristics of (E)GPRS. A. Delayed TBF In order to avoid frequent TBF releases and reestablishments within a talkspurt, the TBF release can be delayed by at least the inter-arrival time of the speech frames. This ensures that consecutive PDUs not exceeding a certain inter-arrival time are transmitted within the same TBF. Thus, frequent TBF releases and re-establishments during a talkspurt are avoided improving the end-to-end delay by 15 ms for low network load situations. This is approximately the time required for the establishment of a TBF, which induces queuing of five speech frames and consequently higher delays for these speech frames. A second benefit of delaying the TBF release is that the amount of signaling previously required for the TBF handling is reduced by a factor of 1 in low-load situations and good radio conditions. Especially in EGPRS, when it is possible to use coding schemes providing data rates up to 59 kbit/s per TS, the RLC sender queue is cleared quickly. That causes a TBF release every 6 th speech frame and a re-establishment of a TBF for consecutive PDUs. The 95 percentile of the uplink end-to-end delay of the speech frames generated by the speech coder G is depicted in Fig. 1, which consists of two curves, one for the unmodified TBF release as described in the standard and one using the delayed TBF mechanism. The end-to-end delay comprises the delay generated by the speech coder and the transmission delay of the radio link. The core network is considered to be ideal, introducing no extra delay. The run of both curves are similar for lower MCSs, whereas for MCSs higher than MCS-3 the delayed TBF mechanism starts to improve the situation. When using MCS- 1 to MCS-2, the transmission capacity is not sufficient to transport the data, which therefore starts to queue up at the RLC protocol level. It is assumed that the MSs have a Multislot Capability of 1, i.e. only one timeslot of the TDMA frame can be used for transmission of data. During talkspurts, the speech coder G generates speech frames at 6.3 kbit/s, which adds up to 12 kbit/s due to RTP, UDP, IP, SNDCP and LLC overhead (assuming ideal header MCS unmodified TBF release Delayed TBF release Fig. 1: 95 Percentile of Uplink End-to-End Delay vs. MCS, Comparison between normal TBF release and delayed TBF release.

3 compression). On the other hand MCS-1 (MCS-2) provides a maximum data rate of 8.8 kbit/s (11.2 kbit/s). MCS-3 is the most suitable coding scheme, when no realtime supporting mechanism is implemented, because the data rate of the speech coder best matches the data rate provided by the (E)GPRS system. Note, that an uplink TBF establishment takes approximately 15 ms which induces queuing of the first 5 speech frames. Thus, the maximum data rate provided by the transmission system should be higher than the data rate of the application (incl. protocol overhead) to be able to decrease the queue size during the talkspurt. On the other hand, if the data rate of the sender is higher than provided by MCS-3, the RLC sender queue is cleared quickly and the TBF release is triggered, which requires a re-establishment of the TBF when new packets enter the queue. When delaying the TBF release, no signaling has to be exchanged within a talkspurt and less queueing on RLC level occurs. B. Semi-Permanent TBF To avoid not only TBF releases during a talkspurt, but also the delay at the beginning of a talkspurt, the semi-permanent TBF mode provides permanent assigned resources. Using the semi-permanent TBF Mode, the TBF is established once at the beginning of a call setup and is released as soon as the call is terminated. Hence, no extra delay is introduced by frequent establishments of the TBF at the beginning and during talkspurts. During inactive periods, the assigned resources are reduced in order to minimize the waste of resources and to be able to multiplex best effort data during the silence periods of the real-time user. By keeping the TBF alive, it becomes necessary to assign TSs periodically and to send USFs to allow an inactive MS to change to the active mode. The slot assignment during inactive periods is configured to schedule resources every 4 ms. The influence of the Multislot Capability is depicted in Fig. 2. It becomes clear, that the data rate provided by a single-slot terminal and a coding scheme with strong forward error protection like MCS-1 to MCS-3, is not sufficient to transport the data in time. End-to-end delays of about 1 ms and less are observed for coding schemes providing high data rates or Multislot capabilities of two or higher. From Fig. 2 we can also see that the delay stays constant although increasing the channel rate. The bandwidth provided by the higher modulation and coding schemes is only partly utilized by the application, which generates data at a constant and fixed data rate. In such cases it would be beneficial to have the possibility of adapting the application, i.e. the voice coder, to the channel conditions and achieve a better voice quality when the channel conditions allow the usage of higher MCSs. On the other hand the capacity available in those conditions could be utilized for the transmission of besteffort data as discussed in section D of this chapter. The influence of the granularity of resource assignments during the inactive period is shown in Fig. 3. A granularity of zero equals a circuit switched mode, where a Packet Data Traffic Channel (PDTCH) is permanently allocated to the 95 Percentile of End-to-end Delay [ms] MCS Single Slot 2 slots 3 slots 4 slots Fig. 2: 95 Percentile of Uplink End-to-end Delay, Semi-permanent TBF mode with different Multislot Capabilities of the Mobile Stations MS. The granularity represents the number of radio blocks (each consisting of 4 bursts) which are skipped in inactive periods, before a radio block is assigned again. Assuming single-slot (E)GPRS terminals, high delays are again observed for MCS-1 to MCS-2, due to the too low data rates. Assuming coding schemes with high data rates (MCS-5 to MCS-9) the (E)GPRS network provides in principle enough capacity to transport the VoIP data stream. As can be seen from Fig. 3 only the permanent assignment of resources yields better performance than the delayed TBF mode explained before. Assigning every second radio block during inactive periods of the mobile station results in similar delay values. These results suggest to prefer the delayed TBF mode as it provides better utilization of the radio resources. The permanent TBF mode, i.e. granularity of zero, and the semi-permanent TBF mode, granularity of one, are again compared in Fig. 4. The graph depicts the impact of the number of active VoIP calls on the end-to-end delay. As only one cell with 8 PDTCH is simulated, the delay rises as soon as the number of VoIP calls exceeds 8. The delay already increases slightly for 8 parallel VoIP calls due to the signaling carried in parallel to the user data. Delays of 95 Percentile of End-to-end Delay [ms] MCS Granularity Granularity 1 Granularity 2 Granularity 3 Fig. 3: 95 Percentile of Uplink End-to-end Delay for the Semipermanent TBF Mode

4 95 Percentile of Uplink End-to-end Delay Number of VoIP calls Semipermanent TBF Permanent TBF Fig. 4: 95 Percentile of Uplink End-to-end Delay, Semi-permanent TBF Mode and Permanent TBF mode (Single-Slot Mobile Stations) approximately 125 ms can be realized using the semipermanent TBF mode. C. Enhanced Permanent TBF The investigations undertaken and described in the previous sections have revealed the principle advantages of the circuit switched approaches, which exhibit low transmission delays, but eliminate the resource utilization gain which pure packet switched approaches show. The optimization approaches followed were aiming at introducing some circuit switched behavior into the packet switched world of (E)GPRS. In the approach described in this section we want to optimize the multiplexing between realtime and best-effort data flows. This approach will help to achieve lower transmission delays for the realtime flows at the expense of the best-effort flows. The aim is to improve the transmission delay of the semi-permanent TBF approach and keep a similar performance level for the multiplexing of besteffort data. On the other hand, when keeping similar delay bounds as in the semi-permanent case the multiplexing performance should be improved, i.e. the resource utilization on the radio link should be improved. As we have seen in section B, the semi-permanent TBF approach exhibits different delay and multiplexing performance depending on the chosen granularity. It is obvious that a lower granularity, i.e. more resources are used to poll for activity of the realtime data-flow, improves the transmission delay for the realtime flow at the expense of resource utilization. The aim is to improve this situation by coupling realtime and best-effort transmission more closely. This has been achieved by defining a special block format to be used on the radio link for the transmission of best-effort data, which is multiplexed into the idle periods of a realtime flow. The multiplexing scheme is depicted in Fig. 5. The assignment of radio resources during active periods of the realtime flow is similar to a static PDTCH assigning the radio resources constantly to the realtime flow (1,5b,6 in Fig. 5). When the realtime application gets idle (speech pause), the resource assignment is changed in such a way that three bursts of a radio block are assigned for the transmission of best effort data and only the remaining single burst is reserved for the realtime flow (3,4,5a). The start of an idle Active RT application Inactive RT application Active RT application a 5b 6 Uplink TSs assigned and used by realtime user Uplink TSs assigned to realtime user but not used Uplink TSs assigned and used by non-realtime user Fig. 5: Multiplexing of realtime and non-realtime traffic period of the realtime applications is detected by a number of consecutive unused radio blocks (2). In principle the same considerations as in the Delayed TBF scheme are applicable for choosing the number of idle radio blocks indicating that the application generating realtime data entered the idle state. In this state best-effort data can be transmitted but instant access to the radio resources is still guaranteed for the realtime flow due to the remaining assignment of a single burst per radio block. When realtime data becomes available again, the single burst can be used to transfer the first data, which will trigger the BSS to switch back to the full assignment of radio resources to the realtime application. Instead of sending true data, one can of course send a dummy packet only, as one burst is wasted anyway in order to keep the radio block alignment. This approach proved to meet the expectations explained earlier in this section. As can be seen from Fig. 6, the uplink delay of realtime data packets improved by 8 ms when configuring the schemes in such a way that the resource utilization is kept the same. This means that the granularity of the semi-permanent mode has to be set to 3, i.e. three data blocks are assigned to the best-effort transmission in between two assignments for the realtime flow. The drawback of this approach is the required modification of the radio block format. CDF Delay [ms] Enhanced Permanent TBF Semi-Permanent TBF with granularity 3 Fig. 6: Uplink delay for the semi-permanent mode and the enhanced semi-permanent mode

5 D. Combining Realtime and Best-Effort Traffic When taking the delay requirements of realtime applications into account and considering the findings of the earlier sections, it is obvious that the number of parallel realtime flows needs to be limited in order to not exceed the delay boundaries. Nevertheless, in such an admission controlled scenario the (E)GPRS system can still provide service to best effort flows having a lower priority than the realtime flows. For the assessment of a shared usage of the (E)GPRS system allowing best-effort traffic to utilize the capacity not consumed by the realtime flows, we apply round-robin scheduling with two priorities. The scheduling is always done per timeslot running the first round-robin cycle over the realtime flows allocated on a timeslot and secondly over the best-effort flows. The second cycle is not invoked when realtime data can be scheduled. The scheduling scheme is depicted in Fig. 8 Despite applying prioritization schemes, the realtime flows get affected when allowing additional best effort data in the system (see Fig. 9). The observed increase of the transmission delay of realtime data is due to the additional signaling messages required for the transmission of the best effort data. These signaling messages are usually prioritized against any user data traffic and thus interfere with the transmission of realtime user data. CDF of uplink end-to-end delay Delay [ms] No. of WWW Processes = No. of WWW Processes = 64 Fig. 7: CDF of Uplink End-to-end Delay of Speech Frames for different Number of WWW best effort user. 1.Priority RR 2.Priority RR TS 1 TS 2 TS 3 TS 4 TS 5 TS 6 TS 7 TS 8 RT RT RT RT RT RT RT RT Fig. 8: Scheduling principle in case of two priority classes. System Throughput [kbit/s] No. of WWW processes No. of VoIP user = 8 No. of VoIP user = Fig. 9: WWW System Throughput With and Without RT Traffic Nevertheless, the additional support for best effort flows seems to be mandatory as a (E)GPRS system applying admission control for the realtime flows might utilize only 5% of its capacity, when aiming at reasonable end-to-end delays for the transmission of the realtime data (see Fig. 9). We observe that adding a sustainable number of realtime flows to a number of best effort-flows would decrease the capacity available for the best effort flows by 5%, i.e. 5% of the capacity would be wasted if no best-effort flows would be in the system. V. CONCLUSIONS In this paper we have presented performance figures for the end-to-end delay of VoIP packets conveyed by an (E)GPRS network. Extensive simulations involving most protocols of the (E)GPRS network have revealed their weaknesses in supporting realtime applications requiring low transmission delays. It has been observed that the procedures and rules defined for the establishment and termination of physical connections increase the end-to-end transmission delay when continuous streams of small application data packets have to be transmitted. Three modifications to the handling of these physical connections have been proposed in this paper to improve the performance. Additionally the transmission delay has been improved by changing the medium access mechanisms of (E)GPRS. The modifications are introduced carefully in order to stay compliant to most EGPRS functions. The investigations have revealed that by applying the proposed modifications the delay requirements for the transmission of VoIP packets can be met. We recommend to use the presented schemes named Delayed TBF Release and Enhanced Permanent TBF to decrease the transmission delay of (E)GPRS networks. Secondly, we recommend to use admission control schemes to limit the number of parallel VoIP sessions in the (E)GPRS network. This limitation guarantees a certain load limit and thus avoids an increase of the end-to-end delay due to queuing in the (E)GPRS network. When setting the limits for the admission control such that the delay requirements of the realtime applications are fulfilled, the radio resources would not be sufficiently

6 utilized. Therefore it is also recommended to allow additional best-effort traffic in order to increase the utilization of the (E)GPRS network. We proved by simulation that such an approach raises the network utilization to an acceptable level without interfering too much with the realtime data flows. VI. REFERENCES [1] 3GPP, Technical Specification Group Services and System Aspects Service Aspects, Architecture for an ALL IP Network (3G TS Version 1..). 3rd Generation Partnership Project, October [2] A. Furuskär et al., EDGE: Enhanced Data Rates for GSM and TDMA/136, Evolution, IEEE Personal Communications, June 1999, Vol. 6, No. 3 [3] 3GPP Technical Specification Group Services and System Aspects Service Aspects, Services and Service Capabilities, TS version May [4] A. Schieder, U. Horn, R. Kalden, Performance analysis of realtime applications in mobile packet switched networks, European Wireless 99, October 6-8, Munich, Germany [5] ITU-T, International Telecommunication Union. Recommendation G.723.1, General Aspects of Digital Transmission Systems, Dual Rate Speech Coder for Multimedia Communications Transmitting at 5.3 kbit/s and 6.3 kbit/s. ITU-T, March [6] ITU-T, International Telecommunication Union. Recommendation H.323, Series H: Audiovisual and Multimedia Systems, Infrastructure of Audiovisual Services System and Terminal Equipment for Audiovisual Services. ITU-T, November [7] ETSI, Digital Cellular Telecommunications System (Phase2+); General Packet Radio Service (GPRS) Mobile Station (MS) Base Station System (BSS) Interface; Radio Link Control/ Medium Access Control (RLC/MAC) Protocol (GSM 4.6 Version 8.2. Release 1999), ETSI, December [8] ETSI, Digital Cellular Telecommunications System (Phase2+); General Packet Radio Service (GPRS); Overall description of the GPRS Radio Interface; Stage 2 (GSM 3.64 version 8.3. Release 1999), ETSI, February 2.