Video Quality Measurement For 3G Handsets

Internet provides many services. VOIP (Voice over IP) is one such service also known as Internet Telephony or IP Telephony. Using VOIP we can make voice telephony calls, participate in video conferences, etc over data networks (WAN'S and LAN'S) or internet. VOIP operates by first converting voice data into digital form, organizing them into packets, transmitting them through the most convenient route to their destination and finally reassembling them at the destination. Protocols like SIP/RTP, H.323, MGCP are designed which perform all the above steps. This project aims to make a video call from a 3G Mobile to an IP phone via Asterisk Gateway. Asterisk to act as bridge for video call between 3G-IP network must capture the audio/video stream from 3G mobile, convert captured stream into an IP compatible stream and send stream to an IP client and vice-versa. Asterisk needs to support AMR codec for audio and MPEG-4 codec for video and H.324M protocol stack for capturing audio/video streams from 3G Mobile. Asterisk currently supports audio codec's like GSM, G.729, A-law, and U-law. It allows H.261, H.263 video streams as pass-through. It supports VOIP protocols like SIP/RTP, MGCP, and H.323 which allows it to interface with other devices. This project aims to implement AMR codec, H.324M protocol stack, MPEG-4, bridging functions between SIP/RTP-ISDN and 3G Mobile in Asterisk which allows a 3G phone to call a SIP client via Asterisk. This thesis discusses the implementation of AMR in asterisk as well as SIP protocol and SIP soft phones

Utilizing DSP for IP telephony applications in mobile terminals

Tässä diplomityössä etsitään ja määritellään optimaalinen ohjelmistoarkkitehtuuri reaaliaikaisen puheenkoodauksen mahdollistamiseksi mobiilin laitteen Internet-puheluohjelmistossa. Arkkitehtuurille asetettiin vaatimus, jonka mukaan puhelu ja siihen liittyvä puheen reaaliaikaisuus ei saa rajoittaa tai liikaa kuormittaa laitteen muuta toiminnallisuutta. Työssä käytetty mobiili laite tarjoaa mahdollisuuden hyödyntää kahta prosessoria. Toinen prosessoreista on tarkoitettu yleisille käyttöjärjestelmille sekä ohjelmistoille ja toinen signaalinkäsittelyoperaatioille. Suunniteltu arkkitehtuuri yhdistää näiden kahden prosessorin toiminnallisuuden ja mahdollistaa reaaliaikaisen puheenkoodauksen (sekä toisto että äänitys) mobiliisissa laitteessa. Arkkitehtuuri toteutettiin ja sen suorituskykyä arvioitiin erilaisilla mittauksilla ja parametreilla. Havaittiin, että toteutus suoriutuu erinomaisesti sille asetetuista vaatimuksista. Todettiin myös, että käytettäessä ainoastaan laitteen yhtä prosessoria reaaliaikavaatimus ei täyty. Tämä johtuu puhekoodekin matemaattisesta kompleksisuudesta ja laitteen rajoitetuista ominaisuuksista. Työn aikana jätettiin kaksi patenttihakemusta.In this thesis, an optimal software architecture is studied and defined for enabling a real-time speech coding scheme in an Internet telephony application of a mobile terminal. According to a requirement set for the architecture, a phone call and the related real-time speech coding shall not limit or overload other functionality of the terminal. The mobile terminal utilized in this thesis provides a potential to take advantage of the efficiency of a dual core processor. One of the processors is designed for general purpose operating systems, and the other one for signal processing operations. The designed software architecture combines the functionality of these processors and enables real-time speech coding (both playback and capture) in the device. The architecture was implemented and its performance was evaluated with different measurements and parameters. It was observed that the implementation outperforms the requirements set. It was also confirmed that the performance of the general purpose processor is inadequate for real-time operations with the chosen speech coder/decoder. Two patent applications were filed by the author during the writing of this thesis

ADAPTIVE SPEECH QUALITY IN VOICE-OVER-IP COMMUNICATIONS

The quality of VoIP communication relies significantly on the network that transports the voice packets because this network does not usually guarantee the available bandwidth, delay, and loss that are critical for real-time voice traffic. The solution proposed here is to manage the voice-over-IP stream dynamically, changing parameters as needed to assure quality. The main objective of this dissertation is to develop an adaptive speech encoding system that can be applied to conventional (telephony-grade) and wideband voice communications. This comprehensive study includes the investigation and development of three key components of the system. First, to manage VoIP quality dynamically, a tool is needed to measure real-time changes in quality. The E-model, which exists for narrowband communication, is extended to a single computational technique that measures speech quality for narrowband and wideband VoIP codecs. This part of the dissertation also develops important theoretical work in the area of wideband telephony. The second system component is a variable speech-encoding algorithm. Although VoIP performance is affected by multiple codecs and network-based factors, only three factors can be managed dynamically: voice payload size, speech compression and jitter buffer management. Using an existing adaptive jitter-buffer algorithm, voice packet-size and compression variation are studied as they affect speech quality under different network conditions. This study explains the relationships among multiple parameters as they affect speech transmission and its resulting quality. Then, based on these two components, the third system component is a novel adaptive-rate control algorithm that establishes the interaction between a VoIP sender and receiver, and manages voice quality in real-time. Simulations demonstrate that the system provides better average voice quality than traditional VoIP

3GPP QoS Model for Networks Using 3GPP QoS Classes

This draft describes an NSIS QoS Model (QOSM) based on 3GPP QoS classes and bearer service attributes. Specifically, this draftdescribes additional optional parameters for QSPEC which carries 3GPP QOSM specific information and how the QSPEC information should be processed in QNEs.\u

Testing of voice quality of VoLTE/VoWiFi technology

Tato diplomová práce se zabývá možnostmi použití ITU-T G.107 E-modelu pro objektivní neintrusivní měření kvality přenosu hlasu v sítích LTE a Wi-Fi. V první části práce jsou prezentovány používané metody měření hlasové kvality, zejména algoritmus POLQA, rozbor E-modelu a přehled technologií VoLTE a VoWiFi. Hlavní částí práce je návrh algoritmu výpočtu R-faktoru, lineárního ukazatele kvality přenosu hlasu, vycházejícího z parametrů měřených uživatelskými zařízení s OS Android. Návrh vychází z rozsáhlých měření technologie VoLTE a jeho úspěšnost predikce hlasové kvality je vyhodnocována na základě srovnání s měřeními algoritmem POLQA. V práci jsou dále uvedeny možnosti implementace algoritmu pro měření hlasové kvality na zařízeních s OS Android.This master’s thesis deals with application of the ITU-T G.107 E-model for objective non-intrusive voice transmission quality measurements in LTE and Wi-Fi networks. The first part presents techniques used for voice quality measurements, particularly algorithm POLQA, analysis of the E-model and overview of the VoLTE and VoWiFi technologies. The main part of this paper consists of design of the R-factor calculation formula using parameters measured by Android OS powered devices. The algorithm design is based on extensive VoLTE measurements and its voice quality prediction successfulness is evaluated by comparison with POLQA measurements. The paper also presents implementation possibilities of the proposed algorithm on devices with Android OS.

TCP-Friendly Rate Control with Token Bucket for VoIP Congestion Control

TCP Friendly Rate Control (TFRC) is a congestion control algorithm that provides a smooth transmission rate for real-time network applications. TFRC refrains from halving the sending rate on every packet drop, instead it is adjusted as a function of the loss rate during a single round trip time. TFRC has been proven to be fair when competing with TCP flows over congested links, but it lacks quality-of-service parameters to improve the performance of real-time traffic. A problem with TFRC is that it uses additive increase to adjust the sending rate during periods with no congestion. This leads to short term congestion that can degrade the quality of voice applications. We propose two changes to TFRC that improve the performance of VoIP applications. Our implementation, TFRC with Token Bucket (TFRC-TB), uses discrete calculated bit rates based on audio codec bandwidth usage to increase the sending rate. Also, it uses a token bucket to control the sending rate during congestion periods. We have used ns2, the network simulator, to compare our implementation to TFRC in a wide range of network conditions. Our results suggest that TFRC-TB can provide a quality of service (QoS) mechanism to voice applications while competing fairly with other traffic over congested links