EVALUATION AND ANALYSIS OF SIP AND VOIP PERFORMANCE WITH PRESENCE TRAFFIC OVER HSPA

Size: px
Start display at page:

Download "EVALUATION AND ANALYSIS OF SIP AND VOIP PERFORMANCE WITH PRESENCE TRAFFIC OVER HSPA"

Transcription

1 EVALUATION AND ANALYSIS OF SIP AND VOIP PERFORMANCE WITH PRESENCE TRAFFIC OVER HSPA Fan Rui¹, Wang Min¹, Mårten Ericson², Stefan Wänstedt² Ericsson Research ¹Beijing, China; ²Luleå, Sweden Abstract This paper investigates the SIP based Presence traffic impact on the VoIP capacity, assuming an HSPA system. The results show that 2000 subscribers per cell with VoIP and presence enabled can be accommodated with a minor capacity loss of 5%. The uplink is the bottleneck for the system. The major reason for the capacity decrease is not the presence packet itself, rather the control channel overhead from DPCCH in UL and A-DPCH in DL due to the channel release delay. The paper investigates different priority based scheduling schemes. The simulations show that the best performance is achieved when VOIP and SIP (including presence and call setup) have equal priority. Prioritizing SIP over VoIP decreases SIP call control setup delay at the expense of lower VoIP capacity. Prioritizing VoIP over SIP or granting Presence the lowest priority results in very large overhead from the control channels, especially the uplink DPCCH. This leads to both decreased VoIP capacity and increased SIP setup delay. However, when UL DPCCH gating is applied, the performance of the scheme to prioritize VoIP over SIP call control and Presence is improved quite a lot and achieves the best trade-off between VoIP capacity and SIP setup delay. Keywords-VoIP; Presence;Capacity; HSPA I. INTRODUCTION IP Multimedia Subsystem (IMS) defined by 3GPP [1] is believed to be the architecture for future communication networks. In an IMS based network, the packet based VoIP service replaces the traditional circuit switched voice service. SIP signaling is used to initiate and release VoIP communication sessions. Presence is a kind of service that enriches the end user perception. Since the network resources are limited, the emergence of Presence traffic is sure to impact SIP call control and VoIP performance, i.e. the SIP call control setup delay and the voice capacity. The objectives of this paper are to investigate the impact of Presence traffic on SIP call control and VoIP performance over HSPA both in the uplink (UL) and in the downlink (DL), and to investigate whether there are some methods to improve the overall SIP call control and VoIP performance in the mixed traffic scenarios. The rest of the paper is organized as follows. Section II shows an introduction of the traffic model and the scheduling schemes considered in this paper. Section III describes the metrics and the simulation assumptions. UL VoIP capacity with theoretical analysis is described in section IV. The simulation results and performance analyses are described in Section V. Finally, discussion and conclusions are presented in Section VI. II. TRAFFIC MODELS AND SCHEDULING SCHEMES A. Traffic Models Three different types of traffic are investigated in this paper: SIP call control traffic, VoIP traffic and Presence traffic. SIP call control traffic is responsible for the setup and release of the VoIP sessions. The SIP message sequence for call setup includes INVITE, 100 Trying, 183 Session in Progress, PRACK, 200 OK, 180 Ringing, 200 OK, and ACK. The SIP sequence for call release includes BYE and 200 OK. The VoIP traffic model is similar to the one used in [2], i.e. AMR 12.2kbps codec rate with 50% voice activity. The session duration follows an exponential distribution with mean value of 30s. Four types of Presence transactions are modeled in this paper: presence update, watcher request, registration and reregistration/de-registration, see Figure 1. Each Presence transaction includes 2-4 Presence messages. The four transactions have intensities of 0.93, 1.34, 0.34 and 0.5/0.18 per user and hour, respectively. The exact intensities of the Presence messages is of course difficult to know, and may vary depending on e.g. user behavior and application. Figure 1. The four types of Presence transactions The Presence traffic, but not SIP call control and VoIP traffic, needs to consider the HS-DSCH channel downswitch timer. This is because generally the HS-DSCH downswitch timer, which is used to control how long a user can occupy resources even when it does not need them, is about 10 times as large as the duration of Presence transactions but less than 1/20 of SIP/VoIP sessions. Until the HS-DSCH down-switch timer expires, DPCCH/F-DPCH channels still exist and consume power even when no Presence messages need to be transmitted /07/$ IEEE

2 The activity of a Presence user is rather low. A Presence user is active only from the time when the first Presence message is transmitted between the user and the Presence server to the time when its channel down-switch timer expires. Assuming the mean duration of one Presence transaction is about 100 ms and the down-switch timer is set to 2 s, the activity factor of one Presence user is approximately ((0.1s + 2s) /3600s)*3.3 = 1.9 merlang where 3.3 is the considered mean number of transactions per user and hour. Assume that all Presence users are VoIP subscribers, the number of Presence users per cell can then be roughly derived from the voice activity per user and the maximum number of active voice users a cell can accommodate. Assuming that a cell typically accommodates up to 50 active voice users and the voice activity per subscriber is 20 merlang, the corresponding number of voice or presence subscribers is then 2500 per cell. In this study the number of Presence subscribers (or users) per cell is varied from 2000 to 8000 as we believe presence subscribers are not necessarily VoIP users. This is done to investigate the effect on SIP call control and VoIP performance. B. Scheduling schemes Scheduling schemes play an important role in HSPA systems. In this paper, EUL employs the so-called nonscheduled mode while a delay sensitive scheduler is deployed in downlink [3]. Non-scheduled mode in uplink means each UE itself decides when and how much data to transmit to the network without any limitation from NodeB. Since different kinds of traffic are mixed in the system, priority based scheduling is natural to be considered here. The prioritization is done between users. Traffic of lower priority is scheduled only when there is no higher priority traffic in the scheduler queues. Depending on whether SIP call control signaling could be distinguished from Presence traffic at the scheduler or not, three priority schemes are considered. SIP call control together with Presence is prioritized over VoIP or vice versa when SIP call control and Presence is indistinguishable. SIP call control is prioritized over VoIP which is in turn prioritized over Presence when SIP call control can be distinguished from Presence. For users belonging to the same traffic type, a delay sensitive scheduler determines which user should be scheduled. C. Simulation models and parameters A detailed dynamic multi-cell HSPA system simulator is employed, see [2]. The voice traffic uses RLC UM while the SIP call control and Presence traffic uses RLC AM. The IP headers of voice packets are compressed using ROHC [4]; the resulting header is 3 bytes. SigComp [5] is used for SIP traffic and the mean compression ratio is 4:1. The HS channel down-switch timer is 2 seconds. Detailed simulation parameters and their values are listed in Table I. TABLE I. Variable Number of cells SIMULATION PARAMETERS. Value 21 cells with 2-D antenna ROHC [bytes] 3 Admission Control Code limitation No No VoIP codec rate [kbps] 12.2 Mean VoIP call duration [s] 30 VoIP activity factor [%] 50 Channel donwswitch timer [s] 2 Poweroffset for VoIP E-DPDCH 0 db for 10 ms TTI, 7 db for 2 [db] ms TTI Poweroffset for Presence E- 7 db for 10 ms TTI, 14 db for DPDCH [db] 2 ms TTI Poweroffset for E-DPCCH -10 db for 10 ms TTI, -3 db for (VoIP/Presence) [db] 2 ms TTI UL DPCCH gating [6], which means UL DPCCH only transmits when necessary, e.g. when data or CQI updates are transmitted in uplink, is also considered in this study. UL DPCCH gating can effectively reduce the UL interference and improve UL VoIP performance as indicated in [6]. While in this paper, it is demonstrated that UL DPCCH gating can benefit the system performance even in downlink as downlink and uplink are coupled with each other in this system. III. SIMULATION METRICS AND ASSUMPTIONS The metrics we care for are VoIP capacity and SIP setup delay. SIP setup delay is defined as the duration from the time when UE sends the INVITE message to the other end until the time when it receives 200 OK for the INVITE message. In the simulation, SIP setup delay is just the round trip delay between UE and RNC, not the end-to-end delay. The VoIP capacity is defined as the offered VoIP traffic load at which 95% of the users are satisfied. A user is satisfied if it is accepted by the system and its packet loss ratio is less than 1% per link. Both uplink capacity and downlink capacity are considered. The offered VoIP load and the capacity figures shown in section IV are relative to the downlink VoIP capacity when no Presence load is offered in the system. The number of Presence enabled users per cell increases from 2000 to 8000 to get the corresponding VoIP capacity and SIP setup delay. IV. THEORETICAL ANALYSIS OF UL VOIP CAPACITY We first investigate how Presence traffic affects UL VoIP capacity with a simple theoretical analysis. The CIR requirement for one user in the UL can be written as γ = γ a + a β + a β ) (1) tot c ( c ed ed ec ec where γ c is the DPCCH CIR, β ed and β ec are the linear power offsets for the E-DPDCH channel and E-DPCCH channel, respectively, compared to the DPCCH CIR. α c, α ed and α ec denote the activity factors for the DPCCH channel, E- DPDCH channel and E-DPCCH channel, respectively.

3 For a VoIP user, activity factors of the E-DPDCH channel and E-DPCCH channel can be estimated by α ed = α ec = α voip * (TTI / AMR Frame interval) * R voip (2) where α voip is the activity factor of voice traffic (0.5 is used, same as in the system simulations), and TTI (length) is either 0.002s or 0.01s, AMR frame interval is 0.02s, and R voip denotes the mean transmission number for one MAC PDU of VoIP. The activity factor of the DPCCH channel is 1. Similarly, the activity factors of E-DPDCH channel and E- DPCCH channel for one Presence user can be estimated by a = a = a TTI R 3600 ed, pres ec, pres λ presence presence presence (3) where λ presence is the number of presence messages per hour, TTI is either 0.002s or 0.01s, and α presence is the activity factor of Presence traffic (here 0.5, since only half of the Presence traffic is transmitted in the UL direction), R presence denotes the mean number of transmissions for one MAC PDU of Presence. As mentioned in section 2, the DPCCH channel will exist until the channel down-switch timer (default 2s) expires for the Presence traffic. Because the duration of the Presence transaction is far less than the channel down-switch time, the influence of the duration of the Presence transaction is ignored in equation 4. Thus, the activity factor of the DPCCH channel for one Presence user is estimated by α c = (channel down switch time ) * (average transaction number per hour)/3600 (4) Given (1)-(4), the required CIR for VoIP and Presence, CIR voip and CIR pres can be calculated. So, the CIR loss per presence user is CIR loss = CIR pres / CIR voip (5) Then, assuming the capacity is linear to the CIR [2], the capacity for the VoIP system with Presence traffic is M = M CIR N (6) original where M original denotes the pure VoIP capacity and N pres is the number of presence users in the cell. The comparison between theoretical UL VoIP capacity and the corresponding simulation results is shown in section V. loss pres significant capacity decrease. The DL VoIP capacity is hardly affected at all, 2-3% capacity decrease, while the UL capacity decrease is around 5-6%. Increasing the Presence load further, up to 6000 Presence users/cell, gives a capacity decrease for the UL of around 15%, and around 3-4% in the DL. Figure 2. Theoretical and simulated results (relative VoIP capacity with variation of Presence traffic load). In UL, basically the same capacity is achieved for both 2ms TTI and 10ms TTI when the offered Presence load is less than 6000 users/cell. But a small (5%) difference is found when the offered Presence load is beyond 6000 users/cell. That is because VoIP users with 2ms TTI reach their maximum power limitation earlier than users with 10ms TTI. Thus higher UL noise rise is generated with 2ms TTI and this affects UL VoIP capacity adversely. We can see from Figure 3 that the UE transmission power with 2ms TTI is larger than that with 10 ms TTI when the Presence load is beyond 6000 users/cell while they are quite similar when the Presence load is less than 6000 users/cell. Figure 2 also validates that the theoretical UL capacity matches the simulation results very well. The parameters used in the theoretical analysis are either input to or output from the simulation. About 2-5% difference is found between the simulation results and the theoretical results. 5% capacity difference occurs only when the Presence traffic load is over 6000 users/cell. V. SIMULATION RESULTS AND PERFORMANCE ANALYSIS A. Simulation results without prioritization In this section, the impact of Presence traffic on SIP call control and VoIP performance in an HSPA system with equal priority for SIP call control, VoIP and Presence traffic flows is shown. The delay sensitive scheduler in [3] is used in the DL, and in the UL non-scheduled mode is used. Figure 2 shows the DL and UL VoIP capacities as a function of the Presence load. The first observation of Figure 2 is that the DL capacity is slightly higher than the UL VoIP capacity without Presence load. This difference is enlarged with the increase of Presence load. The second observation is that the system can handle 2000 Presence users without any Figure 3. Uplink power with different traffic scenarios

4 As expected, higher Presence load results in larger SIP setup delay as shown in Figure 4 and although the UL VoIP capacity with 2 ms TTI is lower than that with 10 ms TTI when the offered Presence load is beyond 6000 users/cell, the SIP setup delay with 2 ms TTI is always smaller than that with 10 ms TTI. Figure 5. DL VoIP capacities with different priority schemes Figure 4. Mean SIP setup delay with 10ms TTI and 2ms TTI. B. Priority based scheduling The idea with priority based scheduling is to take advantage of the fact that Presence is not as time sensitive as VoIP. In this section, three different kinds of priority based downlink scheduling schemes are considered, i.e. SIP call control together with Presence prioritized over VoIP or vice versa, and SIP call control prioritized over VoIP, which in turn is prioritized over Presence. As a reference scheduling without prioritization applied for different traffic types is also considered. The offered SIP/VoIP load shown in the figures below is not the absolute one but relative to DL VoIP capacity when no Presence traffic is offered. The same number of Presence enabled users per cell is offered for all the cases, namely 6000 users/cell. We expect to see a smaller SIP setup delay when SIP call control is prioritized and a higher DL VoIP capacity when VoIP is prioritized. As expected, the results in Figure 5 and Figure 6 show that the scheme to prioritize SIP call control and Presence over VoIP decreases SIP setup delay at the expense of lower voice capacity. However, somewhat unexpectedly is that prioritization of VoIP over SIP call control and Presence degrades both VoIP capacity and SIP setup delay. The scheme that distinguishes SIP call control from Presence traffic does not give good overall performance either. The best trade-off between VoIP capacity and SIP setup delay is achieved with the same priority. Figure 6. SIP setup delay with different priority schemes According to Figures 5 and 6, down-prioritization of Presence (Presence traffic is granted with lower or the lowest priority in the mixed traffic scenarios) is not a good idea. The reason is that the actual concurrently active number of Presence users per cell is not only determined by user arrival rate but also by user departure rate. Low priority to Presence traffic prolongs the transaction duration, decreases the user departure rate, and finally increases the concurrently active number of Presence users in the system. With the increase of active number of Presence users in the system, uplink noise rise goes up, which prolongs the transaction duration further as more retransmission are needed. Since both the uplink noise rise and the power overhead is increased as more active users exist in the system, the performance of VoIP and SIP setup is of course degraded. C. Priority combined with UL DPCCH gating DL scheduling methods combined with UL DPCCH gating are investigated in this subsection. It is already known that with UL gating, the UL VoIP capacity increases quite a lot [6]. Here we consider the effect of UL DPCCH gating on DL VoIP performance. Figure 7 demonstrates that when UL DPCCH gating is enabled and VoIP is prioritized over SIP call control as well as Presence DL VoIP capacity is improved a lot and is larger than the scheme with same priority. Furthermore, the

5 corresponding SIP setup delay when VoIP is prioritized over SIP call control and Presence is reduced at the same time and is now comparable to that using the scheme with same priority as shown in Figure 8. The reason is that UL noise rise is reduced considerably from a very high level to a reasonable level with UL DPCCH gating for the scheme when VoIP is prioritized as shown in Figure 9, which gives relatively good radio link quality. The SIP call setup process and Presence transaction can be finished more quickly, which reduces SIP setup delay and the number of active Presence users in the system. Less number of active Presence users in the system means less power overhead due to F- DPCH, which results in increased VoIP capacity as more power can be allocated to the HS-DSCH channel. Actually in the scheme when SIP call control has the highest priority, both VoIP and Presence can benefit from UL DPCCH gating. UL gating does not have apparent effect on DL VoIP performance with same priority in this case. This is because the UL noise rise is reasonable for this scheme without gating. Figure 7. DL VoIP capacity with UL DPCCH gating Figure 9. UL noise rise comparison (No UL VoIP traffic) VI. CONCLUSIONS This paper investigates an HSPA system with VoIP users, where each VoIP user also has a SIP controlled Presence service. The system simulation results in this paper show that an HSPA system without the so called gating concept [6] can accommodate around 2000 Presence subscribers per cell with a minor capacity loss of around 5%. Increasing the number of Presence subscribers to 6000 per cell gives a capacity loss of 15%. Further, the results show that the UL is a bottleneck for a system with presence service. The major reason for the capacity decrease is not the Presence load itself, rather the control channel overhead from DPCCH in UL and A-DPCH in DL due to the channel release delay. Several priority scheduling schemes are also investigated. The simulations show that the best performance is achieved when VoIP and SIP (including presence and call setup) have equal priority. Prioritizing SIP over VoIP decreases SIP call control setup delay at the expense of lower VoIP capacity. Prioritizing VoIP over SIP or granting SIP Presence the lowest priority results in very large overhead from the control channels, especially the uplink DPCCH. This leads to both decreased VoIP capacity and increased SIP setup delay. A shorter channel release time will improve the Presence performance. However, it is important that other services such as chat and web browsing need to have a relatively long channel release time. UL DPCCH gating can improve the Presence performance substantially. With gating applied the channel overhead decreases. Therefore the best performance is achieved when Presence is down-prioritized over VoIP. Figure 8. SIP setup delay with UL DPCCH gating REFERENCES [1] 3GPP TS , IP Multimedia (IM) Subsystem; Stage 2. [2] Stefan Wanstedt, Marten Ericson, Kristofer Sandlund, Mats Nordberg and Tomas Frankkila, Realization and Performance Evaluation of IMS Multimedia Telephony for HSPA, Proc. of PIMRC'06, Helsinki, Finland, September [3] Mårten Ericson, Stefan Wanstedt, Mixed Traffic HSDPA scheduling Impact on VoIP capacity, to be published in VTC 2007 spring. [4] Bormann et al. RFC3095 RObust Header Compression (ROHC), IETF, [5] Price et al. RFC3320 Signaling Compression (SigComp), IETF 2002 [6] 3GPP TR , Continuous Connectivity for Packet Data Users, 3GPP release 7, 2005

Deployment Aspects for VoIP Services over HSPA Networks

Deployment Aspects for VoIP Services over HSPA Networks Nash Technologies Your partner for world-class custom software solutions & consulting Deployment Aspects for VoIP Services over HSPA Networks Jens Mueckenheim, Enrico Jugl, Thomas Wagner, Michael Link,

More information

Performance Evaluation of VoIP Services using Different CODECs over a UMTS Network

Performance Evaluation of VoIP Services using Different CODECs over a UMTS Network Performance Evaluation of VoIP Services using Different CODECs over a UMTS Network Jianguo Cao School of Electrical and Computer Engineering RMIT University Melbourne, VIC 3000 Australia Email: j.cao@student.rmit.edu.au

More information

Circuit-Switched Voice Services over HSPA

Circuit-Switched Voice Services over HSPA Circuit-Switched Voice Services over HSPA 1 Qualcomm Incorporated, Corporate R&D San Diego, USA Abstract Circuit-Switched (CS) Voice Services over HSPA (CSoHS) was recently introduced for 3GPP WCDMA Release

More information

VoIP Shim for RTP Payload Formats

VoIP Shim for RTP Payload Formats PITALS 50 pt 32 pt VoIP Shim for RTP Payload Formats draft-johansson-avt-rtp-shim Ingemar Johansson, Ericsson AB Outline MTSI in 3GPP Voice service requirements Problems with RTCP Why is inband signaling

More information

Impact of Flexible RLC PDU Size on HSUPA Performance

Impact of Flexible RLC PDU Size on HSUPA Performance Nash Technologies Your partner for world-class custom software solutions & consulting Enrico Jugl, Michael Link, Jens Mueckenheim* *Hochschule Merseburg, Germany Outline Motivation Flexible RLC PDU Size

More information

Packet Scheduling for Voice over IP over HSDPA in Mixed Traffic Scenarios with Different End-to-End Delay Budgets

Packet Scheduling for Voice over IP over HSDPA in Mixed Traffic Scenarios with Different End-to-End Delay Budgets VI INTERNATIONAL TELECOMMUNICATIONS SYMPOSIUM (ITS6), SEPTEMBER 3-6, 6, FORTALEZA-CE, BRAZIL Packet Scheduling for Voice over IP over HSDPA in Mixed Traffic Scenarios with Different End-to-End Delay Budgets

More information

VoIP in 3G Networks: An End-to- End Quality of Service Analysis

VoIP in 3G Networks: An End-to- End Quality of Service Analysis VoIP in 3G etworks: An End-to- End Quality of Service Analysis 1 okia etworks P.O.Box 301, 00045 okia Group, Finland renaud.cuny@nokia.com Renaud Cuny 1, Ari Lakaniemi 2 2 okia Research Center P.O.Box

More information

Priority-Coupling A Semi-Persistent MAC Scheduling Scheme for VoIP Traffic on 3G LTE

Priority-Coupling A Semi-Persistent MAC Scheduling Scheme for VoIP Traffic on 3G LTE Priority-Coupling A Semi-Persistent MAC Scheduling Scheme for VoIP Traffic on 3G LTE S. Saha * and R. Quazi ** * Helsinki University of Technology, Helsinki, Finland ** University of Dhaka, Dhaka, Bangladesh

More information

Providing reliable and efficient VoIP over WCDMA

Providing reliable and efficient VoIP over WCDMA Providing reliable and efficient VoIP over WCDMA Mårten Ericson, Lotta Voigt and Stefan Wänstedt The architecture of the IP Multimedia Subsystem (IMS) defined by the Third Generation Partnership Project

More information

RESOURCE ALLOCATION FOR INTERACTIVE TRAFFIC CLASS OVER GPRS

RESOURCE ALLOCATION FOR INTERACTIVE TRAFFIC CLASS OVER GPRS RESOURCE ALLOCATION FOR INTERACTIVE TRAFFIC CLASS OVER GPRS Edward Nowicki and John Murphy 1 ABSTRACT The General Packet Radio Service (GPRS) is a new bearer service for GSM that greatly simplify wireless

More information

Solution for cell edge performance improvement and dynamic load balancing. Qualcomm Technologies, Inc.

Solution for cell edge performance improvement and dynamic load balancing. Qualcomm Technologies, Inc. HSPA+ Multiflow Solution for cell edge performance improvement and dynamic load balancing Feburary 1, 2014 Qualcomm Technologies, Inc. Not to be used, copied, reproduced, or modified in whole or in part,

More information

Cellular Network Planning and Optimization Part XI: HSDPA. Jyri Hämäläinen, Communications and Networking Department, TKK, 25.1.

Cellular Network Planning and Optimization Part XI: HSDPA. Jyri Hämäläinen, Communications and Networking Department, TKK, 25.1. Cellular Network Planning and Optimization Part XI: HSDPA Jyri Hämäläinen, Communications and Networking Department, TKK, 25.1.2008 HSDPA HSDPA = High Speed Downlink Packet Access. Release 5 was the first

More information

of the existing VoLTE roaming and interconnection architecture. This article compares existing circuit-switched models with the earlier

of the existing VoLTE roaming and interconnection architecture. This article compares existing circuit-switched models with the earlier VoLTE 3GPP Roaming Further Development of LTE/LTE-Advanced LTE Release 10/11 Standardization Trends VoLTE Roaming and ion Standard Technology In 3GPP Release 11, the VoLTE roaming and interconnection architecture

More information

Parallel CS + PS and associated SRB

Parallel CS + PS and associated SRB RAB Example Parallel CS + PS and associated SRB This document details 2 simultaneous RABs and associated SRBs from TS32.8 The RABs are:. Conversational symmetrical Circuit Switched 64 kbps 2. Interactive

More information

Performance Comparison of Control-less Scheduling Policies for VoIP in LTE UL

Performance Comparison of Control-less Scheduling Policies for VoIP in LTE UL Performance Comparison of Control-less Scheduling Policies for VoIP in LTE UL Haiming Wang Nokia Device R&D/Wireless System Research Nokia (China) Investment Corporation Limited, 100176 Beijing, China

More information

VoIP-Kapazität im Relay erweiterten IEEE 802.16 System

VoIP-Kapazität im Relay erweiterten IEEE 802.16 System VoIP-Kapazität im Relay erweiterten IEEE 802.16 System 21. ComNets-Workshop Mobil- und Telekommunikation Dipl.-Ing. Karsten Klagges ComNets Research Group RWTH Aachen University 16. März 2012 Karsten Klagges

More information

1 Introduction 1 1.1 Services and Applications for HSPA 3 1.2 Organization of the Book 6 References 7

1 Introduction 1 1.1 Services and Applications for HSPA 3 1.2 Organization of the Book 6 References 7 Figures and Tables About the Authors Preface Foreword Acknowledgements xi xix xxi xxiii xxv 1 Introduction 1 1.1 Services and Applications for HSPA 3 1.2 Organization of the Book 6 References 7 2 Overview

More information

Performance of UMTS Code Sharing Algorithms in the Presence of Mixed Web, Email and FTP Traffic

Performance of UMTS Code Sharing Algorithms in the Presence of Mixed Web, Email and FTP Traffic Performance of UMTS Code Sharing Algorithms in the Presence of Mixed Web, Email and FTP Traffic Doru Calin, Santosh P. Abraham, Mooi Choo Chuah Abstract The paper presents a performance study of two algorithms

More information

Capacity of VoIP over HSDPA with Frame Bundling

Capacity of VoIP over HSDPA with Frame Bundling Capacity of VoIP over HSDPA with Frame Bundling Yong-Seok Kim Telecommunication Network Business Samsung Electronics Email: ys708.kim@samsung.com Youngheon Kim Telecommunication Network Business Samsung

More information

Performance Issues of TCP and MPEG-4 4 over UMTS

Performance Issues of TCP and MPEG-4 4 over UMTS Performance Issues of TCP and MPEG-4 4 over UMTS Anthony Lo A.Lo@ewi.tudelft.nl 1 Wiskunde end Informatica Outline UMTS Overview TCP and MPEG-4 Performance Summary 2 1 Universal Mobile Telecommunications

More information

Nokia Networks. Voice over LTE (VoLTE) Optimization

Nokia Networks. Voice over LTE (VoLTE) Optimization Nokia Networks Voice over LTE (VoLTE) Optimization Contents 1. Introduction 3 2. VoIP Client Options 5 3. Radio Network Optimization 6 4. Voice Quality Optimization 11 5. Handset Power Consumption Optimization

More information

HSDPA Throughput Performances Using an Experimental HSDPA Transmission System

HSDPA Throughput Performances Using an Experimental HSDPA Transmission System NTT DoCoMo Technical Journal Vol. 6 No.4 HSDPA Throughput Performances Using an Experimental HSDPA Transmission System Shinya Tanaka, Hiroyuki Ishii, Tomoki Sao, Yousuke Iizuka and Takeshi Nakamori The

More information

Enhanced HSDPA Mobility Performance: Quality and Robustness for VoIP Service

Enhanced HSDPA Mobility Performance: Quality and Robustness for VoIP Service Enhanced HSDPA Mobility Performance: Quality and Robustness for VoIP Service Abstract 3GPP has standardized CDMA-based packet-switched air interfaces for downlink and uplink, called High-Speed Downlink

More information

Bluetooth voice and data performance in 802.11 DS WLAN environment

Bluetooth voice and data performance in 802.11 DS WLAN environment 1 (1) Bluetooth voice and data performance in 802.11 DS WLAN environment Abstract In this document, the impact of a 20dBm 802.11 Direct-Sequence WLAN system on a 0dBm Bluetooth link is studied. A typical

More information

Introduction VOIP in an 802.11 Network VOIP 3

Introduction VOIP in an 802.11 Network VOIP 3 Solutions to Performance Problems in VOIP over 802.11 Wireless LAN Wei Wang, Soung C. Liew Presented By Syed Zaidi 1 Outline Introduction VOIP background Problems faced in 802.11 Low VOIP capacity in 802.11

More information

Push-to-talk Over Wireless

Push-to-talk Over Wireless Push-to-talk Over Wireless Is the time right for Push-to-talk? Does it work over GPRS? www.northstream.se Conclusions Push-to-talk is a walkie-talkie-type service implemented over mobile networks. US operator

More information

HO Policies for Combined WLAN/UMTS Networks

HO Policies for Combined WLAN/UMTS Networks HO Policies for Combined WLAN/UMTS Networks Sven Wiethölter Telecommunication Networks Group TU Berlin Telecommunication Networks Group Technische Universität Berlin Project Overview Project partners Goal:

More information

Study of the impact of UMTS Best Effort parameters on QoE of VoIP services

Study of the impact of UMTS Best Effort parameters on QoE of VoIP services Study of the impact of UMTS Best Effort parameters on QoE of VoIP services Jose Oscar Fajardo, Fidel Liberal, Nagore Bilbao Department of Electronics and Telecommunciations, University of the Basque Country

More information

Applicability of UDP-Lite for Voice over IP in UMTS Networks

Applicability of UDP-Lite for Voice over IP in UMTS Networks Applicability of -Lite for Voice over IP in UMTS Networks Frank Mertz, Ulrich Engelke, Peter Vary RWTH Aachen University, Institute of Communication Systems and Data Processing (IND) D-5256 Aachen, Germany

More information

Performance Evaluation of AODV, OLSR Routing Protocol in VOIP Over Ad Hoc

Performance Evaluation of AODV, OLSR Routing Protocol in VOIP Over Ad Hoc (International Journal of Computer Science & Management Studies) Vol. 17, Issue 01 Performance Evaluation of AODV, OLSR Routing Protocol in VOIP Over Ad Hoc Dr. Khalid Hamid Bilal Khartoum, Sudan dr.khalidbilal@hotmail.com

More information

An Improved Mobile VoIP Call Setup with Pre-detection of Callee MS State Information

An Improved Mobile VoIP Call Setup with Pre-detection of Callee MS State Information An Improved Mobile VoIP Call Setup with Pre-detection of MS State Information Soonuk Seol 1 and Sungsoo Cho 2 1 School of Electrical, Electronics & Communication Engineering, Korea University of Technology

More information

Customer Training Catalog Training Programs WCDMA RNP&RNO Technical Training

Customer Training Catalog Training Programs WCDMA RNP&RNO Technical Training Customer Training Catalog Training Programs Customer Training Catalog Training Programs WCDMA RNP&RNO Technical Training HUAWEI Learning Service 2015 COMMERCIAL IN CONFIDENCE 1 Customer Training Catalog

More information

LTE Evolution for Cellular IoT Ericsson & NSN

LTE Evolution for Cellular IoT Ericsson & NSN LTE Evolution for Cellular IoT Ericsson & NSN LTE Evolution for Cellular IoT Overview and introduction White Paper on M2M is geared towards low cost M2M applications Utility (electricity/gas/water) metering

More information

Analysis of QoS parameters of VOIP calls over Wireless Local Area Networks

Analysis of QoS parameters of VOIP calls over Wireless Local Area Networks Analysis of QoS parameters of VOIP calls over Wireless Local Area Networks Ayman Wazwaz, Computer Engineering Department, Palestine Polytechnic University, Hebron, Palestine, aymanw@ppu.edu Duaa sweity

More information

2G/3G Mobile Communication Systems

2G/3G Mobile Communication Systems 2G/3G Mobile Communication Systems Winter 2012/13 Integrated Communication Systems Group Ilmenau University of Technology Outline 2G Review: GSM Services Architecture Protocols Call setup Mobility management

More information

This specification this document to get an official version of this User Network Interface Specification

This specification this document to get an official version of this User Network Interface Specification This specification describes the situation of the Proximus network and services. It will be subject to modifications for corrections or when the network or the services will be modified. Please take into

More information

TECHNICAL CHALLENGES OF VoIP BYPASS

TECHNICAL CHALLENGES OF VoIP BYPASS TECHNICAL CHALLENGES OF VoIP BYPASS Presented by Monica Cultrera VP Software Development Bitek International Inc 23 rd TELELCOMMUNICATION CONFERENCE Agenda 1. Defining VoIP What is VoIP? How to establish

More information

Performance Evaluation of Quality of VoIP Service over UMTS-UTRAN R99

Performance Evaluation of Quality of VoIP Service over UMTS-UTRAN R99 Glasgow, June, 24 th -28 th 2007 Performance Evaluation of Quality of VoIP Service over UMTS-UTRAN R99 Andrea Barbaresi, Andrea Mantovani LAB Contacts: andrea.barbaresi@telecomitalia.it Via G. Reiss Romoli,

More information

HSPA: High Speed Wireless Broadband From HSDPA to HSUPA and beyond. HSPA: High Speed Wireless Broadband From HSDPA to HSUPA and Beyond

HSPA: High Speed Wireless Broadband From HSDPA to HSUPA and beyond. HSPA: High Speed Wireless Broadband From HSDPA to HSUPA and Beyond HSPA: High Speed Wireless Broadband From HSDPA to HSUPA and beyond HSPA: High Speed Wireless Broadband From HSDPA to HSUPA and Beyond Introduction... 3 HSPA Explained... 3 HSPA Technology... 4 HSDPA...4

More information

LTE Performance and Analysis using Atoll Simulation

LTE Performance and Analysis using Atoll Simulation IOSR Journal of Electrical and Electronics Engineering (IOSR-JEEE) e-issn: 2278-1676,p-ISSN: 2320-3331, Volume 9, Issue 6 Ver. III (Nov Dec. 2014), PP 68-72 LTE Performance and Analysis using Atoll Simulation

More information

VoIP over Wireless Opportunities and Challenges

VoIP over Wireless Opportunities and Challenges Prof. Dr. P. Tran-Gia VoIP over Wireless Opportunities and Challenges Universität Würzburg Lehrstuhl für verteilte Systeme H.323 RTP Codec Voice-over-IP over Wireless (VoIPoW) UDP IMS G723.1 SIP G729 HSDPA

More information

LTE Mobility Enhancements

LTE Mobility Enhancements Qualcomm Incorporated February 2010 Table of Contents [1] Introduction... 1 [2] LTE Release 8 Handover Procedures... 2 2.1 Backward Handover... 2 2.2 RLF Handover... 3 2.3 NAS Recovery... 5 [3] LTE Forward

More information

Introduction to VoIP Technology

Introduction to VoIP Technology Lesson 1 Abstract Introduction to VoIP Technology 2012. 01. 06. This first lesson of contains the basic knowledge about the terms and processes concerning the Voice over IP technology. The main goal of

More information

CDMA Network Planning

CDMA Network Planning CDMA Network Planning by AWE Communications GmbH www.awe-com.com Contents Motivation Overview Network Planning Module Air Interface Cell Load Interference Network Simulation Simulation Results by AWE Communications

More information

Architecture Overview NCHU CSE LTE - 1

Architecture Overview NCHU CSE LTE - 1 Architecture Overview NCHU CSE LTE - 1 System Architecture Evolution (SAE) Packet core networks are also evolving to the flat System Architecture Evolution (SAE) architecture. This new architecture optimizes

More information

DISCUSSION. Sophia-Antipolis, France August 16-20, 1999. Agenda item: 14.4 Golden Bridge Technology CPCH Delay Measurements for TS25.

DISCUSSION. Sophia-Antipolis, France August 16-20, 1999. Agenda item: 14.4 Golden Bridge Technology CPCH Delay Measurements for TS25. TSG RAN WG 2#6 Sophia-Antipolis, France August 16-20, 1999 TSGR2#6(99)802 Agenda item: 14.4 Source: Golden Bridge Technology Title: CPCH Delay Measurements for TS25.331, RRC Protocol Document for: Discussion

More information

SIP : Session Initiation Protocol

SIP : Session Initiation Protocol : Session Initiation Protocol EFORT http://www.efort.com (Session Initiation Protocol) as defined in IETF RFC 3261 is a multimedia signaling protocol used for multimedia session establishment, modification

More information

HSDPA Mobile Broadband Data A Smarter Approach to UMTS Downlink Data

HSDPA Mobile Broadband Data A Smarter Approach to UMTS Downlink Data HSDPA Mobile Broadband Data A Smarter Approach to UMTS Downlink Data UMTS mobile wireless systems have enjoyed widespread uptake of high-quality circuit-switched applications like voice and video telephony.

More information

International Journal of Advanced Research in Computer Science and Software Engineering

International Journal of Advanced Research in Computer Science and Software Engineering Volume 2, Issue 11, November 2012 ISSN: 2277 128X International Journal of Advanced Research in Computer Science and Software Engineering Research Paper Available online at: www.ijarcsse.com Automated

More information

VoIP Bandwidth Considerations - design decisions

VoIP Bandwidth Considerations - design decisions VoIP Bandwidth Considerations - design decisions When calculating the bandwidth requirements for a VoIP implementation the two main protocols are: a signalling protocol such as SIP, H.323, SCCP, IAX or

More information

An Introduction to VoIP Protocols

An Introduction to VoIP Protocols An Introduction to VoIP Protocols www.netqos.com Voice over IP (VoIP) offers the vision of a converged network carrying multiple types of traffic (voice, video, and data, to name a few). To carry out this

More information

Extended-rtPS Algorithm for VoIP Services in IEEE 802.16 systems

Extended-rtPS Algorithm for VoIP Services in IEEE 802.16 systems Extended-rtPS Algorithm for VoIP Services in IEEE 802.16 systems Howon Lee, Taesoo Kwon and Dong-Ho Cho Department of Electrical Engineering and Computer Science Korea Advanced Institute of Science and

More information

An Efficient Scheduling Scheme to Enhance the Capacity of VoIP Services in Evolved UTRA Uplink

An Efficient Scheduling Scheme to Enhance the Capacity of VoIP Services in Evolved UTRA Uplink An Efficient Scheduling Scheme to Enhance the Capacity of VoIP Services in Evolved UTRA Uplink Yong-Seok Kim Abstract In this paper, an efficient scheduling scheme is proposed to increase the available

More information

1 Introduction to mobile telecommunications

1 Introduction to mobile telecommunications 1 Introduction to mobile telecommunications Mobile phones were first introduced in the early 1980s. In the succeeding years, the underlying technology has gone through three phases, known as generations.

More information

The future of mobile networking. David Kessens <david.kessens@nsn.com>

The future of mobile networking. David Kessens <david.kessens@nsn.com> The future of mobile networking David Kessens Introduction Current technologies Some real world measurements LTE New wireless technologies Conclusion 2 The future of mobile networking

More information

3G smartphones. ericsson White paper Uen 284 23-3250 February 2015

3G smartphones. ericsson White paper Uen 284 23-3250 February 2015 ericsson White paper Uen 284 23-3250 February 2015 3G smartphones optimizing user experience and network efficiency Rapid global smartphone uptake is creating new mobile data traffic patterns. There is

More information

ANALYSIS OF LONG DISTANCE 3-WAY CONFERENCE CALLING WITH VOIP

ANALYSIS OF LONG DISTANCE 3-WAY CONFERENCE CALLING WITH VOIP ENSC 427: Communication Networks ANALYSIS OF LONG DISTANCE 3-WAY CONFERENCE CALLING WITH VOIP Spring 2010 Final Project Group #6: Gurpal Singh Sandhu Sasan Naderi Claret Ramos (gss7@sfu.ca) (sna14@sfu.ca)

More information

Receiving the IP packets Decoding of the packets Digital-to-analog conversion which reproduces the original voice stream

Receiving the IP packets Decoding of the packets Digital-to-analog conversion which reproduces the original voice stream Article VoIP Introduction Internet telephony refers to communications services voice, fax, SMS, and/or voice-messaging applications that are transported via the internet, rather than the public switched

More information

Clearing the Way for VoIP

Clearing the Way for VoIP Gen2 Ventures White Paper Clearing the Way for VoIP An Alternative to Expensive WAN Upgrades Executive Overview Enterprises have traditionally maintained separate networks for their voice and data traffic.

More information

Implementing SIP and H.323 Signalling as Web Services

Implementing SIP and H.323 Signalling as Web Services Implementing SIP and H.323 Signalling as Web Services Ge Zhang, Markus Hillenbrand University of Kaiserslautern, Department of Computer Science, Postfach 3049, 67653 Kaiserslautern, Germany {gezhang, hillenbr}@informatik.uni-kl.de

More information

Unlicensed Mobile Access (UMA) Handover and Packet Data Performance Analysis

Unlicensed Mobile Access (UMA) Handover and Packet Data Performance Analysis Unlicensed Mobile Access (UMA) Handover and Packet Data Performance Analysis Andres Arjona Nokia Siemens Networks andres.arjona@nsn.com Hannu Verkasalo Helsinki University of Technology hannu.verkasalo@tkk.fi

More information

Priority-Based Congestion Control Algorithm for Cross-Traffic Assistance on LTE Networks

Priority-Based Congestion Control Algorithm for Cross-Traffic Assistance on LTE Networks Priority-Based Congestion Control Algorithm for Cross-Traffic Assistance on LTE Networks Lung-Chih Tung, You Lu, Mario Gerla Department of Computer Science University of California, Los Angeles Los Angeles,

More information

Performance Evaluation of Access Selection Algorithms for VoIP on Wireless Multi-Access Networks

Performance Evaluation of Access Selection Algorithms for VoIP on Wireless Multi-Access Networks VI INTERNATIONAL TELECOMMUNICATIONS SYMPOSIUM (ITS26), SEPTEMBER 3-6, 26, FORTALEZA-CE, BRAZIL Performance Evaluation of Access Selection Algorithms for VoIP on Wireless Multi-Access Networks A. P. da

More information

NSN White paper November 2013. From Voice over IP to Voice over LTE

NSN White paper November 2013. From Voice over IP to Voice over LTE NSN White paper November 2013 From Voice over IP to Voice over LTE CONTENTS 1. Introduction 3 2. VoLTE markets 4 3. VoLTE technology 5 3.1 VoLTE user experience 5 3.1.1 VoLTE talk time 5 3.1.2 VoLTE service

More information

MIMO Antenna Systems in WinProp

MIMO Antenna Systems in WinProp MIMO Antenna Systems in WinProp AWE Communications GmbH Otto-Lilienthal-Str. 36 D-71034 Böblingen mail@awe-communications.com Issue Date Changes V1.0 Nov. 2010 First version of document V2.0 Feb. 2011

More information

point to point and point to multi point calls over IP

point to point and point to multi point calls over IP Helsinki University of Technology Department of Electrical and Communications Engineering Jarkko Kneckt point to point and point to multi point calls over IP Helsinki 27.11.2001 Supervisor: Instructor:

More information

Open IMS Core with VoIP Quality Adaptation

Open IMS Core with VoIP Quality Adaptation Open IMS Core with VoIP Quality Adaptation Is-Haka Mkwawa, Emmanuel Jammeh, Lingfen Sun, Asiya Khan and Emmanuel Ifeachor Centre for Signal Processing and Multimedia Communication School of Computing,Communication

More information

192620010 Mobile & Wireless Networking. Lecture 5: Cellular Systems (UMTS / LTE) (1/2) [Schiller, Section 4.4]

192620010 Mobile & Wireless Networking. Lecture 5: Cellular Systems (UMTS / LTE) (1/2) [Schiller, Section 4.4] 192620010 Mobile & Wireless Networking Lecture 5: Cellular Systems (UMTS / LTE) (1/2) [Schiller, Section 4.4] Geert Heijenk Outline of Lecture 5 Cellular Systems (UMTS / LTE) (1/2) q Evolution of cellular

More information

Delivery of Voice and Text Messages over LTE

Delivery of Voice and Text Messages over LTE Delivery of Voice and Text Messages over LTE 1. The Market for Voice and SMS! 2. Third Party Voice over IP! 3. The IP Multimedia Subsystem! 4. Circuit Switched Fallback! 5. VoLGA LTE was designed as a

More information

Performance Evaluation of Quality of VoIP service over UMTS-UTRAN R99

Performance Evaluation of Quality of VoIP service over UMTS-UTRAN R99 Performance Evaluation of Quality of VoIP service over UMTS-UTRAN R99 Andrea Barbaresi, Andrea Mantovani Telecom Italia - Via G. Reiss Romoli, 274 I-1148 Torino (TO), Italy andrea.barbaresi@telecomitalia.it

More information

ECE 358: Computer Networks. Homework #3. Chapter 5 and 6 Review Questions 1

ECE 358: Computer Networks. Homework #3. Chapter 5 and 6 Review Questions 1 ECE 358: Computer Networks Homework #3 Chapter 5 and 6 Review Questions 1 Chapter 5: The Link Layer P26. Let's consider the operation of a learning switch in the context of a network in which 6 nodes labeled

More information

SIP Trunking and Voice over IP

SIP Trunking and Voice over IP SIP Trunking and Voice over IP Agenda What is SIP Trunking? SIP Signaling How is Voice encoded and transported? What are the Voice over IP Impairments? How is Voice Quality measured? VoIP Technology Confidential

More information

Analysis of IP Network for different Quality of Service

Analysis of IP Network for different Quality of Service 2009 International Symposium on Computing, Communication, and Control (ISCCC 2009) Proc.of CSIT vol.1 (2011) (2011) IACSIT Press, Singapore Analysis of IP Network for different Quality of Service Ajith

More information

Transport Layer Protocols

Transport Layer Protocols Transport Layer Protocols Version. Transport layer performs two main tasks for the application layer by using the network layer. It provides end to end communication between two applications, and implements

More information

REPORT ITU-R M.2134. Requirements related to technical performance for IMT-Advanced radio interface(s)

REPORT ITU-R M.2134. Requirements related to technical performance for IMT-Advanced radio interface(s) Rep. ITU-R M.2134 1 REPORT ITU-R M.2134 Requirements related to technical performance for IMT-Advanced radio interface(s) (2008) TABLE OF CONTENTS... Page 1 Introduction... 2 2 Scope and purpose... 2 3

More information

Simulation of Quality of Service Mechanisms in the UMTS Terrestrial Radio Access Network

Simulation of Quality of Service Mechanisms in the UMTS Terrestrial Radio Access Network Simulation of Quality of Service Mechanisms in the UMTS Terrestrial Radio Access Network A. B. García, M. Álvarez-Campana, E. Vázquez and J. Berrocal Departamento de Ingeniería de Sistemas Telemáticos,

More information

VoIP network planning guide

VoIP network planning guide VoIP network planning guide Document Reference: Volker Schüppel 08.12.2009 1 CONTENT 1 CONTENT... 2 2 SCOPE... 3 3 BANDWIDTH... 4 3.1 Control data 4 3.2 Audio codec 5 3.3 Packet size and protocol overhead

More information

3GPP TS 34.109 V3.10.0 (2004-09)

3GPP TS 34.109 V3.10.0 (2004-09) TS 34.109 V3.10.0 (2004-09) Technical Specification 3rd Generation Partnership Project; Technical Specification Group Radio Access Network; Terminal logical test interface; Special conformance testing

More information

Project Code: SPBX. Project Advisor : Aftab Alam. Project Team: Umair Ashraf 03-1853 (Team Lead) Imran Bashir 02-1658 Khadija Akram 04-0080

Project Code: SPBX. Project Advisor : Aftab Alam. Project Team: Umair Ashraf 03-1853 (Team Lead) Imran Bashir 02-1658 Khadija Akram 04-0080 Test Cases Document VOIP SOFT PBX Project Code: SPBX Project Advisor : Aftab Alam Project Team: Umair Ashraf 03-1853 (Team Lead) Imran Bashir 02-1658 Khadija Akram 04-0080 Submission Date:23-11-2007 SPBX

More information

Computer Networks. Chapter 5 Transport Protocols

Computer Networks. Chapter 5 Transport Protocols Computer Networks Chapter 5 Transport Protocols Transport Protocol Provides end-to-end transport Hides the network details Transport protocol or service (TS) offers: Different types of services QoS Data

More information

3GPP Technologies: Load Balancing Algorithm and InterNetworking

3GPP Technologies: Load Balancing Algorithm and InterNetworking 2014 4th International Conference on Artificial Intelligence with Applications in Engineering and Technology 3GPP Technologies: Load Balancing Algorithm and InterNetworking Belal Abuhaija Faculty of Computers

More information

Technote. SmartNode Quality of Service for VoIP on the Internet Access Link

Technote. SmartNode Quality of Service for VoIP on the Internet Access Link Technote SmartNode Quality of Service for VoIP on the Internet Access Link Applies to the following products SmartNode 1000 Series SmartNode 2000 Series SmartNode 4520 Series Overview Initially designed

More information

Multimedia Communications Voice over IP

Multimedia Communications Voice over IP Multimedia Communications Voice over IP Anandi Giridharan Electrical Communication Engineering, Indian Institute of Science, Bangalore 560012, India Voice over IP (Real time protocols) Internet Telephony

More information

OPPORTUNISTIC SCHEDULING OF VOICE AND DATA TRAFFIC IN WIRELESS NETWORKS. Thomas Bonald and Luca Muscariello

OPPORTUNISTIC SCHEDULING OF VOICE AND DATA TRAFFIC IN WIRELESS NETWORKS. Thomas Bonald and Luca Muscariello EuroFGI Workshop on IP QoS and Traffic Control P. Pereira (Ed.) Lisbon, Portugal, December 6 7, 2007 OPPORTUNISTIC SCHEDULING OF VOICE AND DATA TRAFFIC IN WIRELESS NETWORKS Thomas Bonald and Luca Muscariello

More information

Performance of VoIP Call Set-up Over Satellite-UMTS Using Session Initiation Protocol

Performance of VoIP Call Set-up Over Satellite-UMTS Using Session Initiation Protocol Performance of VoIP Call Set-up Over Satellite-UMTS Using Session Initiation Protocol V. Y. H. Kueh, R. Tafazolli, B. G. Evans Centre for Communication Systems Research (CCSR), University of Surrey, Guildford,

More information

Calculating Bandwidth Requirements

Calculating Bandwidth Requirements Calculating Bandwidth Requirements Codec Bandwidths This topic describes the bandwidth that each codec uses and illustrates its impact on total bandwidth. Bandwidth Implications of Codec 22 One of the

More information

Final for ECE374 05/06/13 Solution!!

Final for ECE374 05/06/13 Solution!! 1 Final for ECE374 05/06/13 Solution!! Instructions: Put your name and student number on each sheet of paper! The exam is closed book. You have 90 minutes to complete the exam. Be a smart exam taker -

More information

An Interference Avoiding Wireless Network Architecture for Coexistence of CDMA 2000 1x EVDO and LTE Systems

An Interference Avoiding Wireless Network Architecture for Coexistence of CDMA 2000 1x EVDO and LTE Systems ICWMC 211 : The Seventh International Conference on Wireless and Mobile Communications An Interference Avoiding Wireless Network Architecture for Coexistence of CDMA 2 1x EVDO and LTE Systems Xinsheng

More information

3 The Network Architecture

3 The Network Architecture SIP-H323: a solution for interworking saving existing architecture G. De Marco 1, S. Loreto 2, G. Sorrentino 3, L. Veltri 3 1 University of Salerno - DIIIE- Via Ponte Don Melillo - 56126 Fisciano(Sa) Italy

More information

This sequence diagram was generated with EventStudio System Designer (http://www.eventhelix.com/eventstudio).

This sequence diagram was generated with EventStudio System Designer (http://www.eventhelix.com/eventstudio). 24-Feb-13 15:23 (Page 1) This call flow describes the call setup from one IMS subscriber to ISUP PSTN termination. The call is routed via the BGCF (Border Gateway Control Function) to the MGCF (Media Gateway

More information

Voice Quality with VoLTE

Voice Quality with VoLTE Matthias Schulist Akos Kezdy Qualcomm Technologies, Inc. Voice Quality with VoLTE 20. ITG Tagung Mobilkommunikation 2015 Qualcomm Engineering Services Support of Network Operators Strong R&D Base End-to-end

More information

LTE VoIP Capacity with Soft Frequency Reuse. Dipl.-Ing. Maciej Mühleisen ComNets TUHH FFV Workshop 15.3.2013

LTE VoIP Capacity with Soft Frequency Reuse. Dipl.-Ing. Maciej Mühleisen ComNets TUHH FFV Workshop 15.3.2013 LTE VoIP Capacity with Soft Frequency Reuse Dipl.-Ing. Maciej Mühleisen ComNets TUHH FFV Workshop 15.3.2013 1 Outline Motivation VoIP Scheduling Soft Frequency Reuse Scheduler Concept Scenario & Results

More information

The GSM and GPRS network T-110.300/301

The GSM and GPRS network T-110.300/301 The GSM and GPRS network T-110.300/301 History The successful analog 1:st generation mobile telephone systems proved that there is a market for mobile telephones ARP (AutoRadioPuhelin) in Finland NMT (Nordic

More information

How QoS differentiation enhances the OTT video streaming experience. Netflix over a QoS enabled

How QoS differentiation enhances the OTT video streaming experience. Netflix over a QoS enabled NSN White paper Netflix over a QoS enabled LTE network February 2013 How QoS differentiation enhances the OTT video streaming experience Netflix over a QoS enabled LTE network 2013 Nokia Solutions and

More information

IxLoad: Advanced VoIP

IxLoad: Advanced VoIP IxLoad: Advanced VoIP IxLoad in a typical configuration simulating SIP endpoints Aptixia IxLoad VoIP is the perfect tool for functional, performance, and stability testing of SIPbased voice over IP (VoIP)

More information

VoIP QoS. Version 1.0. September 4, 2006. AdvancedVoIP.com. sales@advancedvoip.com support@advancedvoip.com. Phone: +1 213 341 1431

VoIP QoS. Version 1.0. September 4, 2006. AdvancedVoIP.com. sales@advancedvoip.com support@advancedvoip.com. Phone: +1 213 341 1431 VoIP QoS Version 1.0 September 4, 2006 AdvancedVoIP.com sales@advancedvoip.com support@advancedvoip.com Phone: +1 213 341 1431 Copyright AdvancedVoIP.com, 1999-2006. All Rights Reserved. No part of this

More information

Implementing VoIP support in a VSAT network based on SoftSwitch integration

Implementing VoIP support in a VSAT network based on SoftSwitch integration Implementing VoIP support in a VSAT network based on SoftSwitch integration Abstract Satellite communications based on geo-synchronous satellites are characterized by a large delay, and high cost of resources.

More information

Interference Management: From Autonomous to Closely Coordinated Approaches

Interference Management: From Autonomous to Closely Coordinated Approaches Technische Universität München Lehrstuhl für Kommunikationsnetze Prof. Dr.-Ing. J. Eberspächer VDE/ITG Fachgruppe 5.2.4 Workshop Darmstadt 2010 Interference Management and Cooperation Strategies in Communication

More information

3GPP Wireless Standard

3GPP Wireless Standard 3GPP Wireless Standard Shishir Pandey School of Technology and Computer Science TIFR, Mumbai April 10, 2009 Shishir Pandey (TIFR) 3GPP Wireless Standard April 10, 2009 1 / 23 3GPP Overview 3GPP : 3rd Generation

More information

II. IEEE802.11e EDCA OVERVIEW

II. IEEE802.11e EDCA OVERVIEW The 18th Annual IEEE International Symposium on Personal, Indoor and Mobile Radio Communications (PIMRC'7) CACITY IMPROVEMENT OF WIRELESS LAN VOIP USING DISTRIBUTED TRANSMISSION SCHEDULING Kei Igarashi,

More information