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

Size: px
Start display at page:

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

Transcription

1 VoIP in 3G etworks: An End-to- End Quality of Service Analysis 1 okia etworks P.O.Box 301, okia Group, Finland Renaud Cuny 1, Ari Lakaniemi 2 2 okia Research Center P.O.Box 407, okia Group, Finland Abstract-- This paper presents the results of a Quality of Service (QoS) study for VoIP service over 3G WCDMA networks. An end-to-end simulation platform has been used for this purpose. The simulations have been run using Adaptive Multi-Rate (AMR) speech codec at 12.2 kbit/s with combination of RTP, UDP and IPv6 protocols. The simulated transmission path includes two radio links (uplink and downlink), connected with a packet switched core network and UTRA Radio Access etworks with several different radio transmission conditions. Furthermore, RObust Header Compression is applied in both radio links. The results include buffering statistics, end-to-end delay estimates, and packet loss statistics. I. ITRODUCTIO During the last few years, the voice over data network services have gained increased popularity. Quick growth of the Internet Protocol (IP) based networks, especially the Internet, has directed a lot of interest towards Voice over IP (VoIP). The VoIP technology has been used in some cases, to replace traditional long-distance telephone technology, for reduced costs for the end-user. aturally to make VoIP infrastructure and services commercially viable, the Quality of Service (QoS) needs to be at least close to the one provided by the Public Switched Telephone etwork (PST). On the other hand, VoIP associated technology will bring to the end user value added services that are currently not available in PST. On the other front, the current development in the cellular radio network technologies are paving the way towards IP capable radio networks. The so called Third Generation (3G) cellular networks, developed and standardized by the Third Generation Partnership Project (3GPP), will provide IP over wireless services, enabling therefore also VoIP. In current cellular systems, e.g. in GSM, the telephony service is based on circuit switched approach. This service is currently highly optimized for transmission of voice, thereby providing good speech quality and good spectral efficiency. However, carrying VoIP will be also possible in 3G WCDMA networks, e.g. 3GPP release 5, and may be of special interest for the mobile network operators for multiple reasons: Firstly, as the bandwidth for individual flows in packet switched domain is not reserved in advance, the multiplexing effects should bring significant capacity savings. Secondly, VoIP service will be supported by the Session Initiation Protocol (SIP), which is a text-based protocol, similar to HTTP and SMTP, for initiating interactive communication sessions between users [1]. Such sessions can include voice, but also e.g. video, chat, interactive games, and virtual reality. Finally, the convergence towards packet switched and IP technology may convince mobile operators to go for solutions that are truly all-ip in order to simplify network interconnection and network management. aturally, for wide end user acceptance and deployment, the VoIP service is required to provide similar perceived voice quality as provided by current highly optimized GSM networks. The challenges for achieving this include typical VoIP related QoS problems, such as packet loss, delay, and delay variation (i.e. jitter), as well as additional overhead brought by the VoIP protocol stack. Therefore the end-to-end VoIP QoS should be studied and evaluated carefully. As an example, it is likely that the packet switched technology, although managed by e.g. Differentiated Services [2], will generate more delay and jitter than the circuit switched technology. Further additional delay and jitter may be caused by the packet segmentation in the radio interface. The end-toend delay is likely to be close to the maximum delay still providing acceptable conversational quality (around ms [3]), extra attention needs to be paid to jitter: too much jitter for a voice stream may be problematic since basic jitter compensation methods may not apply very well or have limited effects. So one important issue to investigate is whether the jitter in 3G networks, will have negative impact on the end-user perceived voice quality. This paper is organized as follows. Section II presents in details the end-to-end VoIP simulator used for this study. Each component of the tool is described in detail. Section III presents the simulation results focusing on packet loss ratio and end-to-end delays. Finally, the conclusion in section IV summarizes the main finding of this study and points out the areas that could be investigated further. II. ED-TO-ED VOIP SIMULATIO Protocols used by the VoIP over 3G can be roughly divided into two categories: signaling related protocols and media related protocols. Although the signalling protocols, such as SIP, are very important part of a VoIP system, in this study we concentrate only on media related protocols and transmission of media data.

2 To run the end-to-end simulations we developed a VoIP speech simulator application for modeling the telephony application and protocol layers from application down to IP and PDCP. The lower layers required for radio link and core network modelling were simulated using external simulation tools and the resulting network conditions were applied in the VoIP speech simulator using error pattern files. The different components of the simulation chain are described in detail in the following subsections. A. Speech application On application level we assumed usage of Adaptive Multi- Rate (AMR) speech codec, which is a mandatory codec for conversational speech services within 3G systems. For all simulation runs we selected usage of AMR 12.2 kbit/s mode with DTX functionality enabled, and employed bandwidth efficient mode of the AMR RTP payload format. This implies that during talk spurts the source generates 32-byte speech payload at 20 ms intervals, while due to DTX during silence periods we will have 7-byte payload carrying Silence Descriptor (SID) frame at 160 ms intervals. We further assumed the typical VoIP protocol stack employing Real-Time Transport Protocol (RTP) encapsulated in User Datagram Protocol (UDP), which is further carried by the IP. The combination of these protocols introduces total of 40 bytes header data when using IP version 4 (IPv4), and bytes header when using IP version 6 (IPv6). We selected IPv6, which has two implications: the size of an IP packet carrying one AMR frame will be either 92 bytes (speech) or 67 bytes (SID), and we need to enable UDP checksum because the IPv6 header does not include a checksum of its own but the most critical fields of the header are covered as part of the UDP pseudo header. Protocol layers below IP follow the 3GPP release 5 specifications, as illustrated in Figure 1. Application E.g., IP, PPP PDCP RLC MAC L1 MS PDCP RLC MAC Relay GTP -U L1 L1 L1 L1 L1 Uu Iu-PS Gn Gi UTRA 3G - SGS 3G - GGS Figure 1 3GPP Protocol stack E.g., IP, PPP B. Robust Header Compression (ROHC) When operating in the bandwidth limited 3G networks it is important to use the radio band as effectively as possible, and header overhead up to 60 bytes can seriously degrade the spectral efficiency of a VoIP service over such link. The Relay RObust Header Compression (ROHC) protocol [4] has been developed to tackle this problem. ROHC provides link-based compression of IP/UDP/RTP headers, in best case down to 1 byte. The effective compression makes use of the fact that majority of the fields in the combined IP/UDP/RTP header either remain constant or introduce constant change throughout a session. However, the maximum compression mentioned above can only be reached when imposing some limitations, a more typical compressed header size would be three or four bytes. The ROHC operation is based on synchronized compression (at the sender site) and decompression (at the receiver site) contexts. The decompression context is initialised by transmitting full IP/UDP/RTP headers in the beginning of the session. Also irregularities in the transmitted stream e.g. by DTX operation or lost packet can introduce compressed headers slightly larger than in the optimal state. In error prone transmission conditions a feedback mechanism is important part of robust compression operation, enabling recovery in case the synchronization between compressor and de-compressor is lost. The ROHC protocol was implemented in our simulator. The ROCH in R-MODE is assumed on both radio links, providing feedback mechanism to enable safe convergence to optimal compression state. We also assume that ROHC Context Identifier is transmitted as a part of the compressed packet. These settings imply that the minimum size of a compressed IP/UDP/RTP header is four bytes. C. Radio network modeling The model for the radio network included the actual radio link, processing in layers below PDCP and access transport in UTRA. The radio link error patterns were prepared using a separate WCDMA system simulator. Three different radio conditions were investigated, introducing frame error rates (FER) of 1%, 3% and 5%. Additionally we also included error-free case in the set of simulation conditions. Different error patterns were prepared for both uplink (UL) and downlink (DL), and the error patterns were obtained from a traced terminal that was moving along a predefined route. For UL radio network we assumed processing and transport delay of 36 ms, and for DL radio network the corresponding delay is 49 ms. ote that 36+49=85 ms is the lower limit for the time before the ROHC compressor can receive a feedback message from the decompressor regarding a specific packet. This delay is significant in such a way that in beginning of a stream the ROHC decompressor context needs to be initialized by sending full headers, which will be sent until a feedback message indicating successful decompressor context initialization is received. A similar situation can occur also if the decompression context gets corrupted for some reason, e.g. hard handover or excessive amount of transmission errors. However, for this work we assumed that no ROHC decompressor context re-initialization is required during a session.

3 D. RLC, MAC and PDCP layers The WCDMA unacknowledged radio mode is the natural choice for transmitting the VoIP packets over the radio link. This mode provides possibility for segmentation and padding of IP packets into radio Time Transfer Intervals (TTIs) to make best possible usage of allocated radio resources. The radio bearer was configured 16 kbit/s; with TTI length of 20ms this enables transmission 40 bytes of user data at 20ms intervals. E. Packet switched domain modelling In the packet switched domain we considered the following delay components: Delay in the IP backbone, delay in the gateway elements (SGS and GGS) and delay in IuPS interface. Typically, the backbone elements (IP routers) and gateways may introduce some jitter to VoIP traffic, depending on the load in the network. However appropriate traffic prioritisation (e.g. based on Differentiated Services) can limit the queuing delay (and thus potential jitter) to specific values defined by the operator. We modelled this kind of PS domain structure to generate a delay distribution file for a stream of packets transmitted at 20 ms intervals. The resulting delay distribution is illustrated in Figure 2, and it introduces 19 ms average delay with 1.0 ms standard deviation. The minimum and maximum values for the delay are 12.4 ms and 23.7 ms, respectively. Figure 2: Delay distribution in PS domain. F. Buffering Typically an audio playout device in the receiving terminal is synchronized to a local clock signal to make sure that there is always signal available for playback. In practice this implies that a new frame is required regularly at intervals determined by the frame rate. On the other hand, due to jitter the packets can arrive at the receiver at irregular rate that is not synchronous to the playout. Therefore, the buffering of speech packets is needed to ensure continuous data flow between asynchronous input and synchronous output. In VoIP this kind of jitter buffering plays an important part in the overall speech quality. The basic approach to jitter buffering is to wait for a predetermined time after the reception of the first packet before playing out the frame carried by this packet. The purpose of the playout delay is to allow some variation in the arrival times of subsequent packets. Frames arriving after their scheduled playout time are discarded and in the speech decoder point of view they are lost frames. aturally in this approach the predetermined buffering delay is the most important factor of the buffering performance: too short buffering delay will risk buffer underflows when packets do not arrive in time due to jitter, and on the other hand too long buffering time introduces unnecessarily long delay and can also introduce buffer overflows. However, for this study we configured the jitter buffer in receiving terminal in such a way that no frames were discarded, neither due to late arrival nor due to buffer overflow. The main reason for this choice was the aim to concentrate on the QoS issues that are dependent on the network. When considering VoIP traffic over a wireless 3G network, it is not sufficient to buffer only in the receiving terminal. Actually in this environment the most critical link between asynchronous input and synchronous output is between the PS core network and the DL radio network. At this point of data path the units we are buffering are IP packets received from the packet switched core network, which will be forwarded to the radio path. Here we assume a slightly different buffering strategy as described above for jitter buffering in the receiving terminal: instead of relying on long enough buffering delay we use FIFO buffer with limited size (as number of packets in the buffer) and specify a maximum time a packet can be stored in a buffer. I.e. if a predetermined number of packets are already stored in the buffer, a new incoming packet will dropped. And if a packet has been waiting in the buffer for longer time than specified by the discard timer, it will be dropped to avoid accumulating delay for subsequent packets. However, to make sure that the large packets required for ROHC initialization will get through without unfeasibly large value for discard timer we made the assumption that a (tail of a) packet that has been already partially transmitted due to segmentation is never dropped even if the timer has elapsed. Although in general it might not seem sensible to perform buffering in the transmitting terminal for a VoIP application, due to strictly limited radio bandwidth, allocated according to optimally compressed headers, and ROHC initialisation requiring transmission of full IP/UDP/RTP headers, we need to consider also buffering prior to UL transmission. We apply similar buffering mechanism as described for DL, i.e. we specify fixed size FIFO buffer with a discard timer to make sure that this bottleneck does not cause unfeasibly long delay.

4 G. Additional simulation settings We used the same speech input sequence for all simulation runs. This speech sequence has approximately 6 minutes 30 seconds duration and it is an excerpt of a real discussion, and therefore introduces realistic structure of alternating talk spurts and silence periods. The speech is in Finnish and it is recorded in low-noise office environment. The observed speech activity is approximately 50%. We also repeated all simulation scenarios ten times with different randomly selected starting points in the radio link error pattern files and in the PS domain delay distribution file to make sure that the results are not affected by some local anomaly in the simulated network conditions. III. SIMULATIO RESULTS Since a fixed-delay jitter buffering scheme was assumed in the receiving terminal, the total end-to-end delay is fixed throughout the session. However, because of the TTI structure, packet segmentation at the RLC level and ROHC behaviour the network delay on packet level is not fixed throughout the connection. Packets too large to be carried by a single TTI need to be segmented over several TTIs thus introducing longer transmission delay over a radio link, in most cases in both uplink and downlink. Furthermore, some of the subsequent packets following the large packets are also segmented over two TTIs although in principle they could fit into single TTI because they are not aligned with the TTI structure. The reason for this is that when these packets are obtained from the buffer, there is still some room in the tail of the current radio frame, and as much data as possible from the beginning of the next packet, if available, are carried here. For these scenarios there are two causes for frame losses (packet losses); a packet can be lost on the radio path due to transmission errors, or a packet can be dropped due to buffering, either in transmitting terminal, in DL RC or in receiving terminal. We would like to point out one observation regarding frame losses: the observed packet loss rate in the radio link seems to be slightly higher than the nominal frame error rate specified for the error patterns over all radio FER conditions. Because of the segmentation a loss of single radio frame can cause loss of two packets: when a radio frame carrying data from two separate packets is lost, both these packets will be unusable and will be dropped by the receiver. The simulation results are summarized in Table 1. The results include packet loss rate (PLR) and buffering time statistics, as well as end-to-end delays in different scenarios. There is also a further breakdown of packet loss statistics into losses due to DL buffering and losses due to transmission errors on the radio path. Since we carry one AMR frame per packet the FER at speech decoder input equals PLR. ote that losses in the UL terminal buffering are not presented in the table, but they are included in the total packet loss rate. ote also that the average network delay includes the jitter buffering time in the receiving terminal. Radio link FER PLR in DL buff Table 1: Simulation results. PLR on radio Total PLR Avg.DL buff delay Avg. network delay 0% 0.02% 0% 0.06% 9.79ms ms 1% 0.02% 2.05% 2.08% 9.79ms ms 3% 0.02% 6.02% 6.08% 9.79ms ms 5% 0.02% 10.26% 10.31% 9.79ms ms The overall frame error rate (FER) can be used as a rough objective speech quality estimate. Typically, with AMR codec the speech quality can be still considered good when FER is around 1-2%, but it should be noted that also the distribution of frame losses has an effect on the subjective speech quality. IV. COCLUSIOS Our end-to-end Quality of Service analysis shows that 3GPP networks will be able to offer an adequate level of quality for Voice over IP (VoIP) services. The difference in QoS with current voice services technology (CS voice) is very small: The additional packet loss ration introduced by packet switched characteristics is less than 1%, whereas the end-toend network average delay is expected to be around 220ms. The enabling features for the obtained quality level are summarized below: WCDMA unacknowledged mode in radio ROHC at the PDCP layer that allows usage of limited bandwidth radio bearer (16kbits/s). Relevant buffering limits and discarding rules in the PDCP buffer (DL) and in the transmitting terminal to avoid potential cumulative delay and jitter. Differentiated Services support in the core network and backbone to ensure minimal buffering delay in packet switched domain. evertheless there are few other important aspects that require further investigations in order to determine if VoIP services will be quickly deployed in 3G networks. 1. The User Equipment (UE) may contribute to the mouthto-ear delay: The processing time needed to compress the VoIP headers should not be negligible. Also, because the first few packets during the ROHC initialisation phase are transmitted with full headers, the UE may require special buffering mechanism in order to minimize the delay. 2. The radio capacity needed to transfer VoIP flows is slightly higher than the capacity needed for sending circuit

5 switched voice frames, even with header compression. A detailed analysis, that would take pricing into account, would be useful to determine if offering VoIP services in 3G networks is efficient and interesting from an operator business perspective. ACKOWLEDGEMETS The authors wish to thank Zhi-chun Honkasalo and Mattias Wahlqvist for their frequent feedback along this study. Mika Kolehmainen and Outi Hiironniemi also contributed to this work by providing support for radio link error and PS domain delay modelling. REFERECES [1] IETF Session Initiation Protocol (SIP) Working Group, [2] IETF Differentiated Services (DiffServ) Working Group, [3] ITU-T Recommendation G.114, One-way transmission time, 05/2000 [4] RFC 3095, RObust Header Compression (ROHC); Framework and four profiles: RTP, UDP, ESP, and uncompressed, July 2001

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

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

Encapsulating Voice in IP Packets

Encapsulating Voice in IP Packets Encapsulating Voice in IP Packets Major VoIP Protocols This topic defines the major VoIP protocols and matches them with the seven layers of the OSI model. Major VoIP Protocols 15 The major VoIP protocols

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

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

Requirements of Voice in an IP Internetwork

Requirements of Voice in an IP Internetwork Requirements of Voice in an IP Internetwork Real-Time Voice in a Best-Effort IP Internetwork This topic lists problems associated with implementation of real-time voice traffic in a best-effort IP internetwork.

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

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

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

ALCATEL CRC Antwerpen Fr. Wellesplein 1 B-2018 Antwerpen +32/3/240.8550; Suresh.Leroy@alcatel.be +32/3/240.7830; Guy.Reyniers@alcatel.

ALCATEL CRC Antwerpen Fr. Wellesplein 1 B-2018 Antwerpen +32/3/240.8550; Suresh.Leroy@alcatel.be +32/3/240.7830; Guy.Reyniers@alcatel. Contact: ALCATEL CRC Antwerpen Fr. Wellesplein 1 B-2018 Antwerpen +32/3/240.8550; Suresh.Leroy@alcatel.be +32/3/240.7830; Guy.Reyniers@alcatel.be Voice over (Vo) was developed at some universities to diminish

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

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

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

Broadband Networks. Prof. Dr. Abhay Karandikar. Electrical Engineering Department. Indian Institute of Technology, Bombay. Lecture - 29.

Broadband Networks. Prof. Dr. Abhay Karandikar. Electrical Engineering Department. Indian Institute of Technology, Bombay. Lecture - 29. Broadband Networks Prof. Dr. Abhay Karandikar Electrical Engineering Department Indian Institute of Technology, Bombay Lecture - 29 Voice over IP So, today we will discuss about voice over IP and internet

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

Network administrators must be aware that delay exists, and then design their network to bring end-to-end delay within acceptable limits.

Network administrators must be aware that delay exists, and then design their network to bring end-to-end delay within acceptable limits. Delay Need for a Delay Budget The end-to-end delay in a VoIP network is known as the delay budget. Network administrators must design a network to operate within an acceptable delay budget. This topic

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

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

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

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

3. Simulator Description. Figure 1: UMTS Architecture (air interface and radio access network). the data stored at the buffer up to a certain maximum

3. Simulator Description. Figure 1: UMTS Architecture (air interface and radio access network). the data stored at the buffer up to a certain maximum Optimal Radio Acess Bearer Configuration for Voice over in 3G UMTS networks Xavier Pérez-Costa, Albert Banchs 1, Juan Noguera and Sebastià Sallent-Ribes NEC Network Laboratories, Heidelberg, Germany Universitat

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

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

QoS issues in Voice over IP

QoS issues in Voice over IP COMP9333 Advance Computer Networks Mini Conference QoS issues in Voice over IP Student ID: 3058224 Student ID: 3043237 Student ID: 3036281 Student ID: 3025715 QoS issues in Voice over IP Abstract: This

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

VoIP Bandwidth Calculation

VoIP Bandwidth Calculation VoIP Bandwidth Calculation AI0106A VoIP Bandwidth Calculation Executive Summary Calculating how much bandwidth a Voice over IP call occupies can feel a bit like trying to answer the question; How elastic

More information

Indepth Voice over IP and SIP Networking Course

Indepth Voice over IP and SIP Networking Course Introduction SIP is fast becoming the Voice over IP protocol of choice. During this 3-day course delegates will examine SIP technology and architecture and learn how a functioning VoIP service can be established.

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

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

TDM services over IP networks

TDM services over IP networks Keyur Parikh Junius Kim TDM services over IP networks 1. ABSTRACT Time Division Multiplexing (TDM) circuits have been the backbone of communications over the past several decades. These circuits which

More information

W H I T E PA P E R. The concept of robust header compression, ROHC

W H I T E PA P E R. The concept of robust header compression, ROHC The concept of robust header compression, ROHC F E B R U A R Y 2 0 0 4 W W W. E F F N E T. C O M The concept of robust header compression, ROHC C O N T E N T S The need for IP header compression.....................3

More information

Application Note How To Determine Bandwidth Requirements

Application Note How To Determine Bandwidth Requirements Application Note How To Determine Bandwidth Requirements 08 July 2008 Bandwidth Table of Contents 1 BANDWIDTH REQUIREMENTS... 1 1.1 VOICE REQUIREMENTS... 1 1.1.1 Calculating VoIP Bandwidth... 2 2 VOIP

More information

VOICE OVER IP AND NETWORK CONVERGENCE

VOICE OVER IP AND NETWORK CONVERGENCE POZNAN UNIVE RSITY OF TE CHNOLOGY ACADE MIC JOURNALS No 80 Electrical Engineering 2014 Assaid O. SHAROUN* VOICE OVER IP AND NETWORK CONVERGENCE As the IP network was primarily designed to carry data, it

More information

4:3 VoIP Transmission in Fixed Wireless Access networks based on DECT Packet Radio Service

4:3 VoIP Transmission in Fixed Wireless Access networks based on DECT Packet Radio Service 4:3 VoIP Transmission in Fixed Wireless Access networks based on DECT Packet Radio Service Luis M. Contreras Francisco Escrihuela Alcatel España, Radio Communications Division Ramirez de Prado, 5 28045

More information

Choosing the Right Audio Codecs for VoIP over cdma2000 Networks:

Choosing the Right Audio Codecs for VoIP over cdma2000 Networks: Choosing the Right Audio Codecs for VoIP over cdma2000 Networks: System capacity, Voice quality, Delay, and Transcoding issues Dr. Sassan Ahmadi NOKIA Inc. sassan.ahmadi@nokia.com February 8, 2005 1 2005

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

STANDPOINT FOR QUALITY-OF-SERVICE MEASUREMENT

STANDPOINT FOR QUALITY-OF-SERVICE MEASUREMENT STANDPOINT FOR QUALITY-OF-SERVICE MEASUREMENT 1. TIMING ACCURACY The accurate multi-point measurements require accurate synchronization of clocks of the measurement devices. If for example time stamps

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

Network Simulation Traffic, Paths and Impairment

Network Simulation Traffic, Paths and Impairment Network Simulation Traffic, Paths and Impairment Summary Network simulation software and hardware appliances can emulate networks and network hardware. Wide Area Network (WAN) emulation, by simulating

More information

Overview of Voice Over Internet Protocol

Overview of Voice Over Internet Protocol Overview of Voice Over Internet Protocol Purva R. Rajkotia, Samsung Electronics November 4,2004 Overview of Voice Over Internet Protocol Presentation Outline History of VoIP What is VoIP? Components of

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

Unit 23. RTP, VoIP. Shyam Parekh

Unit 23. RTP, VoIP. Shyam Parekh Unit 23 RTP, VoIP Shyam Parekh Contents: Real-time Transport Protocol (RTP) Purpose Protocol Stack RTP Header Real-time Transport Control Protocol (RTCP) Voice over IP (VoIP) Motivation H.323 SIP VoIP

More information

VoIP over MANET (VoMAN): QoS & Performance Analysis of Routing Protocols for Different Audio Codecs

VoIP over MANET (VoMAN): QoS & Performance Analysis of Routing Protocols for Different Audio Codecs VoIP over MANET (VoMAN): QoS & Performance Analysis of Routing Protocols for Different Audio Codecs Said El brak Mohammed Bouhorma Anouar A.Boudhir ABSTRACT Voice over IP (VoIP) has become a popular Internet

More information

Advanced Networking Voice over IP: RTP/RTCP The transport layer

Advanced Networking Voice over IP: RTP/RTCP The transport layer Advanced Networking Voice over IP: RTP/RTCP The transport layer Renato Lo Cigno Requirements For Real-Time Transmission Need to emulate conventional telephone system Isochronous output timing same with

More information

Nortel - 920-803. Technology Standards and Protocol for IP Telephony Solutions

Nortel - 920-803. Technology Standards and Protocol for IP Telephony Solutions 1 Nortel - 920-803 Technology Standards and Protocol for IP Telephony Solutions QUESTION: 1 To achieve the QoS necessary to deliver voice between two points on a Frame Relay network, which two items are

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

Voice over IP. Presentation Outline. Objectives

Voice over IP. Presentation Outline. Objectives Voice over IP Professor Richard Harris Presentation Outline Brief overview of VoIP and applications Challenges of VoIP IP Support for Voice Protocols used for VoIP (current views) RTP RTCP RSVP H.323 Semester

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

5. DEPLOYMENT ISSUES Having described the fundamentals of VoIP and underlying IP infrastructure, let s address deployment issues.

5. DEPLOYMENT ISSUES Having described the fundamentals of VoIP and underlying IP infrastructure, let s address deployment issues. 5. DEPLOYMENT ISSUES Having described the fundamentals of VoIP and underlying IP infrastructure, let s address deployment issues. 5.1 LEGACY INTEGRATION In most cases, enterprises own legacy PBX systems,

More information

Digital Audio and Video Data

Digital Audio and Video Data Multimedia Networking Reading: Sections 3.1.2, 3.3, 4.5, and 6.5 CS-375: Computer Networks Dr. Thomas C. Bressoud 1 Digital Audio and Video Data 2 Challenges for Media Streaming Large volume of data Each

More information

VOICE over IP H.323 Advanced Computer Network SS2005 Presenter : Vu Thi Anh Nguyet

VOICE over IP H.323 Advanced Computer Network SS2005 Presenter : Vu Thi Anh Nguyet VOICE over IP H.323 Advanced Computer Network SS2005 Presenter : Vu Thi Anh Nguyet 1 Outlines 1. Introduction 2. QoS in VoIP 3. H323 4. Signalling in VoIP 5. Conclusions 2 1. Introduction to VoIP Voice

More information

AERONAUTICAL COMMUNICATIONS PANEL (ACP) ATN and IP

AERONAUTICAL COMMUNICATIONS PANEL (ACP) ATN and IP AERONAUTICAL COMMUNICATIONS PANEL (ACP) Working Group I - 7 th Meeting Móntreal, Canada 2 6 June 2008 Agenda Item x : ATN and IP Information Paper Presented by Naoki Kanada Electronic Navigation Research

More information

Voice over IP: RTP/RTCP The transport layer

Voice over IP: RTP/RTCP The transport layer Advanced Networking Voice over IP: /RTCP The transport layer Renato Lo Cigno Requirements For Real-Time Transmission Need to emulate conventional telephone system Isochronous output timing same with input

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

Management of Telecommunication Networks. Prof. Dr. Aleksandar Tsenov akz@tu-sofia.bg

Management of Telecommunication Networks. Prof. Dr. Aleksandar Tsenov akz@tu-sofia.bg Management of Telecommunication Networks Prof. Dr. Aleksandar Tsenov akz@tu-sofia.bg Part 1 Quality of Services I QoS Definition ISO 9000 defines quality as the degree to which a set of inherent characteristics

More information

Transport and Network Layer

Transport and Network Layer Transport and Network Layer 1 Introduction Responsible for moving messages from end-to-end in a network Closely tied together TCP/IP: most commonly used protocol o Used in Internet o Compatible with a

More information

Introduction to VoIP. 陳 懷 恩 博 士 副 教 授 兼 所 長 國 立 宜 蘭 大 學 資 訊 工 程 研 究 所 Email: wechen@niu.edu.tw TEL: 03-9357400 # 255

Introduction to VoIP. 陳 懷 恩 博 士 副 教 授 兼 所 長 國 立 宜 蘭 大 學 資 訊 工 程 研 究 所 Email: wechen@niu.edu.tw TEL: 03-9357400 # 255 Introduction to VoIP 陳 懷 恩 博 士 副 教 授 兼 所 長 國 立 宜 蘭 大 學 資 訊 工 程 研 究 所 Email: wechen@niu.edu.tw TEL: 3-93574 # 55 Outline Introduction VoIP Call Tpyes VoIP Equipments Speech and Codecs Transport Protocols

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

EXPERIMENTAL STUDY FOR QUALITY OF SERVICE IN VOICE OVER IP

EXPERIMENTAL STUDY FOR QUALITY OF SERVICE IN VOICE OVER IP Scientific Bulletin of the Electrical Engineering Faculty Year 11 No. 2 (16) ISSN 1843-6188 EXPERIMENTAL STUDY FOR QUALITY OF SERVICE IN VOICE OVER IP Emil DIACONU 1, Gabriel PREDUŞCĂ 2, Denisa CÎRCIUMĂRESCU

More information

Voice and Fax/Modem transmission in VoIP networks

Voice and Fax/Modem transmission in VoIP networks Voice and Fax/Modem transmission in VoIP networks Martin Brand A1Telekom Austria ETSI 2011. All rights reserved Name : Martin Brand Position: Senior IT Specialist at A1 Telekom Vice Chairman ETSI TC INT

More information

Voice over IP (VoIP) Overview. Introduction. David Feiner ACN 2004. Introduction VoIP & QoS H.323 SIP Comparison of H.323 and SIP Examples

Voice over IP (VoIP) Overview. Introduction. David Feiner ACN 2004. Introduction VoIP & QoS H.323 SIP Comparison of H.323 and SIP Examples Voice over IP (VoIP) David Feiner ACN 2004 Overview Introduction VoIP & QoS H.323 SIP Comparison of H.323 and SIP Examples Introduction Voice Calls are transmitted over Packet Switched Network instead

More information

Cisco Networks (ONT) 2006 Cisco Systems, Inc. All rights reserved.

Cisco Networks (ONT) 2006 Cisco Systems, Inc. All rights reserved. Optimizing Converged Cisco Networks (ONT) reserved. Lesson 2.4: Calculating Bandwidth Requirements for VoIP reserved. Objectives Describe factors influencing encapsulation overhead and bandwidth requirements

More information

12 Quality of Service (QoS)

12 Quality of Service (QoS) Burapha University ก Department of Computer Science 12 Quality of Service (QoS) Quality of Service Best Effort, Integrated Service, Differentiated Service Factors that affect the QoS Ver. 0.1 :, prajaks@buu.ac.th

More information

Quality of Service. Traditional Nonconverged Network. Traditional data traffic characteristics:

Quality of Service. Traditional Nonconverged Network. Traditional data traffic characteristics: Quality of Service 1 Traditional Nonconverged Network Traditional data traffic characteristics: Bursty data flow FIFO access Not overly time-sensitive; delays OK Brief outages are survivable 2 1 Converged

More information

Quality of Service Testing in the VoIP Environment

Quality of Service Testing in the VoIP Environment Whitepaper Quality of Service Testing in the VoIP Environment Carrying voice traffic over the Internet rather than the traditional public telephone network has revolutionized communications. Initially,

More information

Streaming Audio and Video

Streaming Audio and Video Streaming Audio and Video CS 360 Internet Programming Daniel Zappala Brigham Young University Computer Science Department Streaming Audio and Video Daniel Zappala 1/27 Types of Streaming stored audio and

More information

Understanding Latency in IP Telephony

Understanding Latency in IP Telephony Understanding Latency in IP Telephony By Alan Percy, Senior Sales Engineer Brooktrout Technology, Inc. 410 First Avenue Needham, MA 02494 Phone: (781) 449-4100 Fax: (781) 449-9009 Internet: www.brooktrout.com

More information

Voice over IP. Overview. What is VoIP and how it works. Reduction of voice quality. Quality of Service for VoIP

Voice over IP. Overview. What is VoIP and how it works. Reduction of voice quality. Quality of Service for VoIP Voice over IP Andreas Mettis University of Cyprus November 23, 2004 Overview What is VoIP and how it works. Reduction of voice quality. Quality of Service for VoIP 1 VoIP VoIP (voice over IP - that is,

More information

Sync & Sense Enabled Adaptive Packetization VoIP

Sync & Sense Enabled Adaptive Packetization VoIP Sync & Sense Enabled Adaptive Packetization VoIP by Boonchai Ngamwongwattana B.Eng., King Mongkut s Institute of Technology, Ladkrabang, Thailand, 1994 M.S., Telecommunications, University of Pittsburgh,

More information

Combining Voice over IP with Policy-Based Quality of Service

Combining Voice over IP with Policy-Based Quality of Service TechBrief Extreme Networks Introduction Combining Voice over IP with Policy-Based Quality of Service Businesses have traditionally maintained separate voice and data networks. A key reason for this is

More information

AN ANALYSIS OF DELAY OF SMALL IP PACKETS IN CELLULAR DATA NETWORKS

AN ANALYSIS OF DELAY OF SMALL IP PACKETS IN CELLULAR DATA NETWORKS AN ANALYSIS OF DELAY OF SMALL IP PACKETS IN CELLULAR DATA NETWORKS Hubert GRAJA, Philip PERRY and John MURPHY Performance Engineering Laboratory, School of Electronic Engineering, Dublin City University,

More information

IP-Telephony Quality of Service (QoS)

IP-Telephony Quality of Service (QoS) IP-Telephony Quality of Service (QoS) Bernard Hammer Siemens AG, Munich Siemens AG 2001 1 Presentation Outline End-to-end OoS of VoIP services Quality of speech codecs Network-QoS IntServ RSVP DiffServ

More information

Evaluating Data Networks for Voice Readiness

Evaluating Data Networks for Voice Readiness Evaluating Data Networks for Voice Readiness by John Q. Walker and Jeff Hicks NetIQ Corporation Contents Introduction... 2 Determining Readiness... 2 Follow-on Steps... 7 Summary... 7 Our focus is on organizations

More information

VoIP in 802.11. Mika Nupponen. S-72.333 Postgraduate Course in Radio Communications 06/04/2004 1

VoIP in 802.11. Mika Nupponen. S-72.333 Postgraduate Course in Radio Communications 06/04/2004 1 VoIP in 802.11 Mika Nupponen S-72.333 Postgraduate Course in Radio Communications 06/04/2004 1 Contents Introduction VoIP & WLAN Admission Control for VoIP Traffic in WLAN Voice services in IEEE 802.11

More information

Comparison of Voice over IP with circuit switching techniques

Comparison of Voice over IP with circuit switching techniques Comparison of Voice over IP with circuit switching techniques Author Richard Sinden Richard Sinden 1 of 9 Abstract Voice-over-IP is a growing technology. Companies are beginning to consider commercial

More information

VoIP QoS on low speed links

VoIP QoS on low speed links Ivana Pezelj Croatian Academic and Research Network - CARNet J. Marohni a bb 0 Zagreb, Croatia Ivana.Pezelj@CARNet.hr QoS on low speed links Julije Ožegovi Faculty of Electrical Engineering, Mechanical

More information

Computer Networks CS321

Computer Networks CS321 Computer Networks CS321 Dr. Ramana I.I.T Jodhpur Dr. Ramana ( I.I.T Jodhpur ) Computer Networks CS321 1 / 22 Outline of the Lectures 1 Introduction OSI Reference Model Internet Protocol Performance Metrics

More information

MINIMUM NETWORK REQUIREMENTS 1. REQUIREMENTS SUMMARY... 1

MINIMUM NETWORK REQUIREMENTS 1. REQUIREMENTS SUMMARY... 1 Table of Contents 1. REQUIREMENTS SUMMARY... 1 2. REQUIREMENTS DETAIL... 2 2.1 DHCP SERVER... 2 2.2 DNS SERVER... 2 2.3 FIREWALLS... 3 2.4 NETWORK ADDRESS TRANSLATION... 4 2.5 APPLICATION LAYER GATEWAY...

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

920-803 - technology standards and protocol for ip telephony solutions

920-803 - technology standards and protocol for ip telephony solutions 920-803 - technology standards and protocol for ip telephony solutions 1. Which CODEC delivers the greatest compression? A. B. 711 C. D. 723.1 E. F. 726 G. H. 729 I. J. 729A Answer: C 2. To achieve the

More information

IP-based Mobility Management for a Distributed Radio Access Network Architecture. helmut.becker@siemens.com

IP-based Mobility Management for a Distributed Radio Access Network Architecture. helmut.becker@siemens.com IP-based Mobility Management for a Distributed Radio Access Network Architecture helmut.becker@siemens.com Outline - Definition IP-based Mobility Management for a Distributed RAN Architecture Page 2 Siemens

More information

QoS Measurements Methods and Tools

QoS Measurements Methods and Tools QoS Measurements Methods and Tools Contact: Jarmo prokkola Jarmo.prokkola@vtt.fi Tel: +358 20 722 2346 VTT Technical Reseach Centre of Finland Easy Wireless Workshop, IST Summit, Budapest, 05.07.2007 Network

More information

QOS Requirements and Service Level Agreements. LECTURE 4 Lecturer: Associate Professor A.S. Eremenko

QOS Requirements and Service Level Agreements. LECTURE 4 Lecturer: Associate Professor A.S. Eremenko QOS Requirements and Service Level Agreements LECTURE 4 Lecturer: Associate Professor A.S. Eremenko Application SLA Requirements Different applications have different SLA requirements; the impact that

More information

Distributed Systems 3. Network Quality of Service (QoS)

Distributed Systems 3. Network Quality of Service (QoS) Distributed Systems 3. Network Quality of Service (QoS) Paul Krzyzanowski pxk@cs.rutgers.edu 1 What factors matter for network performance? Bandwidth (bit rate) Average number of bits per second through

More information

Optimizing Converged Cisco Networks (ONT)

Optimizing Converged Cisco Networks (ONT) Optimizing Converged Cisco Networks (ONT) Module 2: Cisco VoIP Implementations (Deploy) Calculating Bandwidth Requirements for VoIP Objectives Describe factors influencing encapsulation overhead and bandwidth

More information

The Fax on IP Networks. White Paper February 2011

The Fax on IP Networks. White Paper February 2011 The Fax on IP Networks White Paper February 2011 2 The Fax on IP Networks Contents Overview... 3 Group 3 Fax Technology... 4 G.711 Fax Pass-Through... 5 T.38 IP Fax Relay... 6 Network Design Considerations...

More information

RTP Performance Enhancing Proxy

RTP Performance Enhancing Proxy PACE RTP Performance Enhancing Proxy V2 Whilst the above information has been prepared by Inmarsat in good faith, and all reasonable efforts have been made to ensure its accuracy, Inmarsat makes no warranty

More information

IP QoS Interoperability Issues

IP QoS Interoperability Issues SP-030371 IP QoS Interoperability Issues Source: Contact: SBC Communications, BT Randolph Wohlert Randolph_Wohlert@labs.sbc.com 2 Industry Trend: IP Based Services Next Generation Networks Multi-service

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

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

Voice over Internet Protocol (VoIP) systems can be built up in numerous forms and these systems include mobile units, conferencing units and

Voice over Internet Protocol (VoIP) systems can be built up in numerous forms and these systems include mobile units, conferencing units and 1.1 Background Voice over Internet Protocol (VoIP) is a technology that allows users to make telephone calls using a broadband Internet connection instead of an analog phone line. VoIP holds great promise

More information

3GPP LTE Packet Data Convergence Protocol (PDCP) Sub Layer

3GPP LTE Packet Data Convergence Protocol (PDCP) Sub Layer 3GPP LTE Packet Data Convergence Protocol (PDCP) Sub Layer 2009 Inc. All Rights Reserved. LTE PDCP Sub Layer Functions Header compression and decompression with ROHC Transfer of data and PDCP sequence

More information

Voice over IP (VoIP) for Telephony. Advantages of VoIP Migration for SMBs BLACK BOX. 724-746-5500 blackbox.com

Voice over IP (VoIP) for Telephony. Advantages of VoIP Migration for SMBs BLACK BOX. 724-746-5500 blackbox.com Voice over IP (VoIP) for Telephony Advantages of VoIP Migration for SMBs BLACK BOX Hybrid PBX VoIP Gateways SIP Phones Headsets 724-746-5500 blackbox.com Table of Contents Introduction...3 About Voice

More information

Measuring Data and VoIP Traffic in WiMAX Networks

Measuring Data and VoIP Traffic in WiMAX Networks JOURNAL OF TELECOMMUNICATIONS, VOLUME 2, ISSUE 1, APRIL 2010 Measuring Data and VoIP Traffic in WiMAX Networks 1 Iwan Adhicandra Abstract Due to its large coverage area, low cost of deployment and high

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

QoS Tools in the WAN. Need for QoS on WAN Links. Need for QoS in the WAN

QoS Tools in the WAN. Need for QoS on WAN Links. Need for QoS in the WAN QoS Tools in the WAN Need for QoS on WAN Links This topic defines the need for QoS in a WAN. Need for QoS in the WAN Voice must compete with data. Voice is real-time and must be sent first. Overhead should

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

ENSC 427: COMMUNICATION NETWORKS ANALYSIS ON VOIP USING OPNET

ENSC 427: COMMUNICATION NETWORKS ANALYSIS ON VOIP USING OPNET ENSC 427: COMMUNICATION NETWORKS ANALYSIS ON VOIP USING OPNET FINAL PROJECT Benson Lam 301005441 btl2@sfu.ca Winfield Zhao 200138485 wzhao@sfu.ca Mincong Luo 301039612 mla22@sfu.ca Data: April 05, 2009

More information

Agilent Technologies Performing Pre-VoIP Network Assessments. Application Note 1402

Agilent Technologies Performing Pre-VoIP Network Assessments. Application Note 1402 Agilent Technologies Performing Pre-VoIP Network Assessments Application Note 1402 Issues with VoIP Network Performance Voice is more than just an IP network application. It is a fundamental business and

More information

TCP in Wireless Networks

TCP in Wireless Networks Outline Lecture 10 TCP Performance and QoS in Wireless s TCP Performance in wireless networks TCP performance in asymmetric networks WAP Kurose-Ross: Chapter 3, 6.8 On-line: TCP over Wireless Systems Problems

More information