Miikka Lundan

SIP call setup delay in 3G networks

In this paper, call setup time in Session Initiation Protocol (SIP)-based videotelephony is analyzed. We used a 3G network emulator to measure post-dialing delay, answer-signal delay and call-release delay. The results are compared to local, national, international and overseas intranet LAN calls. Furthermore, we have also studied the effect SIP calls over lossy channels with restricted bandwidth, typical of mobile network signaling bearers.

Human Perception of Lip Synchronization in Mobile Environment

In literature, many studies about human perception of lip synchronization refer to experiences based on TV sets. In these cases, researchers give some hints on how easily humans percept lip synchronization problems. The in-sync region is typically known to be in the range -80ms to +80ms. Within this range most of the test candidates does not detect any lip synchronization skew. The out-of-synch region typically falls outside the range between -160ms and +160ms, where nearly everybody detects the lip synchronization error. Between those boundaries there is a transient area, where the detection of any lip synchronization problem depends on speaker size (head, shoulder or body view). These earlier assumptions based on TV environment do not hold for mobile environment, where the viewing characteristics and of the device itself are different. Therefore, further research is needed. The purpose of this paper is to find new results for human perception of lip synchronization in mobile environment. Results will show that new lip synchronization thresholds apply when using mobile phones and mobile media concepts.

Optimal 3GPP packet-switched streaming service (PSS) over GPRS networks

Abstract3GPP packet-switched streaming service (PSS) is a standardized packet-based mobile streaming service, which is based on IETF RTSP/SDP standards. PSS can be implemented over GPRS networks; however these cannot usually guarantee any data rates or delay bounds, but allow sufficient bandwidth for mobile streaming. GPRS cell reselections pose additional challenges for streaming, since several seconds of data transmission breaks may occur, and data may even be lost. The level of error protection of GPRS is good enough for mobile streaming, if the correct quality of service (QoS) profile is configured. In this paper, we study the effect of different QoS parameters configurations to find optimal values for PSS over GPRS. The paper shows also a method for optimizing cell reselection management at the application layer, in order to provide seamless mobility. Results show that despite all the limitations of a GPRS environment, PSS is feasible with a decent quality of service.

AMR mode selection enhancement in 3G networks

This paper describes methods for mode selection in multirate speech codecs, such as the AMR (Adaptive Multi-Rate), that is the mandatory speech codec selected in 3GPP (3rd Generation Partnership Project) defined mobile networks. Originally, the multirate functionality has been developed for coping with changing radio conditions. The algorithms described in this paper find applicability in IP-based mobile networks, where speech encoded data is encapsulated using the RTP (Real Time Protocol). The main advantages offered by these techniques are improved speech quality and congestion control along the network path between two mobile terminals.

Streaming over EGPRS

Enhanced general packet radio service (EGPRS) is a new technology that provides higher bit rates and better error protection for packet-switched data transmissions, compared to the traditional GPRS technology. Especially video streaming which requires higher bitrates in order to achieve sharp image quality and fluent motion. GPRS allowed the introduction of streaming service with adequate quality, but better technology is needed in order to provide a more enjoyable service. The purpose of this article is to show what new benefits EGPRS brings to packet-switched streaming service (PSS).