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When using Scheduler A and B in this study, the parameterminAgeThd is set to

a static value of 30 ms. Since the use of RTTI reduces the transmission time of a radio block to 10 ms, setting this parameter to 30 ms makes a scheduled user wait until three other users have been scheduled until being scheduled again. 30 ms is therefore optimized for multiplexing four users on the same timeslot bin which results in a total of sixteen users on eight TSs. In a future study, it would be interesting to let this parameter be set dynamically depending on the level of multiplexing. When no multiplexing is present, the value should be set to zero and then increase by 10 ms for each added level of multiplexing. Obviously, there is a limit where the wait time starts affecting the VoIP performance too much.

The simulations in this study were to be run with a static number of users which means that no calls would have been initiated or terminated except at the very beginning and at the end of the simulations. This is obviously far from real life behavior and therefore a dynamic flow of user arrivals and departures would be preferred when running simulations of this kind. This would impact the VoIP capacity even more due to increased SIP signaling payload.

It could also be interesting to investigate how different user types could affect the system capacity. For example, it would be interesting to know what impact users only utilizing the presence service would have on the system. Since a session initiation has to be done for each sent presence message (if only utilizing the pres- ence service), the choice of flow release time would probably affect the performance a great deal.

Furthermore, the human hearing sense can compensate for occasional packet losses, but when several consecutive packets are lost the perceived speech quality is decreased. Therefore, it might be interesting to evaluate not only the overall packet loss rates, but also the number of consecutively lost packets.

Bibliography

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URL: http://www.3gpp.org/ftp/Specs/archive/23_series/23.107/23107-

630.zip [2008-10-18], 2005.

[2] 3GPP. Ts 44.065 mobile station (ms) - serving gprs support node

(sgsn); subnetwork dependent convergence protocol (sndcp) v8.0.0. URL: http://www.3gpp.org/ftp/Specs/archive/44_series/44.065/44065-800.zip [2009-01-30], 2005.

[3] 3GPP. Ts 45.002 multiplexing and multiple access on the radio path v7.7.0.

URL: http://www.3gpp.org/ftp/Specs/archive/45_series/45.002/45002-

770.zip [2008-11-02], 2008.

[4] Ericsson AB. Gprs system survey. LZU 108 876 R3A, 2003. [5] Ericsson AB. Gsm system survey. EN/LZU 108 852 R5A, 2003.

[6] Ericsson AB. The evolution of edge - white paper. URL:

http://www.ericsson.com/technology/whitepapers/3107_The_evolution_of _EDGE_A.pdf [2008-10-11], 2007.

[7] Andreas Bergström. Voip over edge evolution - single session performance and service coverage. Ericsson Confidential Report: EAB-07:067819 Uen, 2007. [8] Per Synnergren & Jan Christoffersson. Analytical delay calculation of the

short ims session setup proposal using wcdma and edge. Ericsson Confidential Report: EAB-05:041425 Uen, 2005.

[9] IETF. Rfc 3261 sip: Session initiation protocol. URL:

http://tools.ietf.org/html/rfc3261 [2008-10-16], 2002.

[10] IETF. Rfc 3262 reliability of provisional response sin the session initiation protocol (sip). URL: http://tools.ietf.org/html/rfc3262 [2008-10-25], 2002. [11] IETF. Rfc 3263 session initiation protocol (sip): Locating sip servers. URL:

http://tools.ietf.org/html/rfc3263 [2008-10-16], 2002.

[12] IETF. Rfc 3265 session initiation protocol (sip)-specific event notification. URL: http://tools.ietf.org/html/rfc3265 [2008-10-16], 2002.

[13] IETF. Rfc 3903 session initiation protocol (sip) extension for event state publication. URL: http://tools.ietf.org/html/rfc3903 [2008-10-15], 2002. [14] IETF. Rfc 3856 a presence event package for the session initiation protocol

(sip). URL: http://tools.ietf.org/html/rfc3856 [2008-10-15], 2004.

[15] Jan Christoffersson & Tomas Jönsson. Presence traffic impact on gsm/edge capacity. Ericsson Limited Internal: EAB -04:013005 Uen, 2005.

[16] Fan Rui & Wang Min. Impact of presence traffic on downlink voip system performance. Ericsson Confidential Technical Report: EAB-06:030370 Uen. [17] Daniel Puaca & Sofia Svedevall. Support for multiple qos conversational users

in gsm mixed traffic. Ericsson Internal Technical Report: EAB/FJG-08:1915 Uen (Ongoing study), 2009.

[18] Ubiquity. Understanding sip - white paper. URL:

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[19] Mats Wernersson. Scheduling of multi-media over 3gpp lte. Ericsson Internal Report: EAB-07:011403 Uen, 2007.

[20] Mats Wernersson. Effects of sip setup signaling on voip performance in lte. Ericsson Wide Internal Report: EAB-07:039951 Uen, 2008.

Appendix A

Abbreviations

AMR Adaptive Multi Rate

BSC Base Station Controller

BSS Base Station System

BTS Base Transceiver Station

CIR Carrier to Interference Ratio

CS Circuit Switched

GGSN Gateway GPRS Support Node

GMM/SM GPRS Mobility Management and Session Management

GPRS General Packet Radio Service

GSM Global Service for Mobile transmission

IMS IP Multimedia Subsystem

IRA Interference Rejection Algorithm

LA Location Area

MCS Modulation and Coding Scheme

MMTel MultiMedia Telephony

MS Mobile Station

MSC Mobile Services Switching Center

PS Packet Switched

PSTN Public Switched Telephone Network

QoS Quality of Service

SGSN Serving GPRS Support Node

SIP Session Initiation Protocol

SMS Short Message Service

TDMA Time Division Multiplex Access

TS Timeslot

UA SIP User Agent

VLR Visitor Location Register

VoIP Voice over IP

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