1-4244-0216-6/06/$20.00 ©2006 IEEE

Mobile Multimedia Group Conferencing - Enriching H.264-based Video by Mobile Source

Specific Multicast Communication Hans L. Cycon, Thomas C. Schmidt, Matthias Wählisch, Mark Palkow and Henrik Regensburg

Abstract — In this paper we report on a multimedia communication software with a distributed architecture and its applications. It is a simple, ready-to-use scheme for distributed presenting, recording and streaming multimedia content over unicast or multicast networks. Dedicated end-to-end bandwidth management optimizes its network resource consumption. Additionally we report on group communications schemes for future applications within mobile networks. We present straightforward extensions to session signalling and source specific multicast routing for transforming (morphing) previous delivery trees into optimal trees rooted at a relocated source. This extension scheme only requires basic signalling mechanisms, explicit joins and prunes, which are present in current multicast routing protocols such as PIM-SM. First evaluations grounded on real-world Internet topologies indicate network performance superior to traditional distribution schemes*.

Index Terms — Video and Multimedia Group Conferencing,

E-learning, Mobile IPv6, Source Specific Multicast Mobility

I. INTRODUCTION

Mobile internet usage becomes more and more a day-to-

day application. Additionally visual devices performing

synchronous communication such as voice or video

conferencing over IP (VoIP/VCoIP) are now almost

ubiquitous. This raises new challenges for the Internet

infrastructure, such as mobile conference users. The

availability of new, truly mobile IP enabled sub network

layers not only offers connectivity to nomadic users at

roaming devices, preserving communication sessions

beyond IP subnet changes, but re-raises questions

concerning the quality of IP service: The constant bit rate

scenarios of voice and video conferencing will appear

significantly disturbed by packet loss intervals, delays or

* Hans L. Cycon is with the FB 1, FHTW Berlin, Treskowalle 8, 10318 Berlin, Germany (e-mail: hcycon@fhtw-berlin.de).

Thomas C. Schmidt is with the Department of Electrical Engineering and Computer Science, HAW Hamburg, Berliner Tor 7, 20099 Hamburg. He is also with the Computer Center, FHTW Berlin, Treskowallee 8, 10318

Berlin, Germany (e-mail: schmidt@fhtw-berlin.de). Matthias Wählisch is with the Computer Center, FHTW Berlin,

Treskowallee 8, 10318 Berlin, Germany (e-mail: mw@fhtw-berlin.de).

Mark Palkow is with the daViKo GmbH Berlin, Hoenower Strasse 35/PF 16, 10318 Berlin, Germany (e-mail: palkow@daviko.com)

Henrik Regensburg is with the FB 1, FHTW Berlin, Treskowalle 8,

10318 Berlin, Germany (e-mail: h.r@fhtw-berlin.de).

jitter exceeding 100 ms. Thus, when heading towards

VCoIP as a standard Internet service, important steps for

global usability have to be taken, focusing on ease and

quality.

In VCoIP conferencing scenarios each member

commonly simultaneously acts as a group listener and a

source. Therefore group communication and especially IP

multicasting will be of particular importance to mobile

environments, where users commonly share frequency

bands of limited capacities. While mobility of a listener can

relatively easily be handled, source mobility presents a

severe problem for multicast packet distribution.

There are two Internet approaches dealing with multicast

group communication: Source Specific Multicast (SSM)

[11] and Any Source Multicast (ASM) [12]. SSM, still in its

design process, is considered a promising improvement of

group distribution techniques. In contrast to ASM, optimal

multicast source trees are constructed immediately from

(S,G), i.e. Source address - Group address router states

subscriptions at the client side, without utilizing network

flooding or Rendezvous Points. Source addresses are to be

acquired by out of band channels, i.e. a SIP [9] session

initiation in conferencing scenarios. As a consequence,

routing simplifies significantly. In VCoIP conferencing

scenarios each member commonly simultaneously acts as a

group listener and a sender. In a mobile scenario, routing

thus invalidates with changing source addresses. Up until

now SSM source mobility remains as an unsolved problem

[17, 18].

In this paper we present a multimedia communication

system including a VCoIP (Video Conferencing over IP)

software with a distributed architecture and some

applications. We further on discuss session mobility with the

special focus on real-time multicast.-

The paper is organized as follows. Section II presents the

basic video conference software and some of its applications

and test results. In Section III we review the basic problems

of multicast source mobility and related work, sketch our

approach to SIP-based source specific group initiation and

introduce tree morphing, our new approach to source

specific multicast sender mobility. Finally, section IV is

dedicated to conclusions and outlook.

II. A DISTRIBUTED COMMUNICATION SYSTEM

A. The Basic Software

The digital audio-visual conferencing system we use is a

server-less multipoint video conferencing software without

MCU developed by the authors [2]. It has been designed in a

peer-to-peer model as a lightweight Internet conferencing

tool aimed at email-like friendliness of use. Guided by the

latter principle, it refrained from implementing H.323 client

requirements [1].

The system is built upon a fast H.264/MPEG-4 AVC

standard conformal video codec implementation [3]. It is a

Baseline profile implementation, optimized for real-time

decoding and encoding by several accelerating measures

like diamond shape motion search, MMX enhanced SAD

motion estimation, fast mode selection and a fast subpel

search strategy.

There is also an application-tailored fast wavelet-based

video codec [4] used for higher available data rate.

By controlling the coding parameters appropriately, the

software permits scaling in bit rate from 24 to 1440 kbit/s on

the fly. Audio data is compressed using a 16 kHz-speech-

optimized variable bit rate codec [5] with extremely short

latencies of 40 ms (plus network packet delay). All streams

can be transmitted by unicast as well as by multicast

protocol. Audio streams are prioritized above video since

audio communication is more sensitive to distortions in

erroneous networks.

An application-sharing facility is included for

collaboration and teleteaching. It enables participants to

share or broadcast not only static documents, but also any

selected dynamic PC action like animations including mouse

pointer movements. All audio/video (A/V) - streams

including dynamic application sharing actions can be

recorded on any site. These data can be displayed locally or

automatically converted into a web streaming format, which

is internet wide available.

This system is equally well suited to intranet and wireless

video conferencing on a best effort basis, since the

audio/video quality can be controlled to adapt the data

stream to the available bandwidth. In faulty network

conditions like poor WLAN links we use unicast TCP

transmissions to avoid distortions by packet loss. For point-

to-multipoint situations like virtual classrooms there is also

a possibility to switch to multicast network transmission via

UDP to minimize computation and transmission load for

networks and senders. To avoid problems with non-routable

private network segments we use Simple Traversal of UDP

over NATs (STUN). This protocol is mostly used for

assisting devices behind a NAT, a firewall or router with

port blocking, see [16].

The joined use of high bandwidth UDP traffic with TCP

updates bound to real-time demands is known to suffer from

distortions due to TCP traffic suppression. Application

sharing in conferencing applications thus is endangered to

encounter disruptions in the event of network congestion.

For a service enhanced synchronous use of UDP media

sessions and application sharing with reliable data transport

requirements, we implemented end-to-end load balancing

employing proprietary extensions to UDP, reliable (RUDP).

We work on its packet identifiers to control application

sharing data flows. On the occasion of a significant amount

(e.g., 5) of unacknowledged packets from shared

application, we slow down video packet transmission to

reserve required resources for real-time application updates,

see [8]. Audio communication remains undisturbed of load-

balancing actions.

Global connection between conference participants in the

internet will be established by a dynamic user session

recording. We denote this by User Session Locator (USL)

and store appropriate session information within a

Lightweight Directory Access Protocol (LDAP) directory

server (see [6] for further details).

B. Application Scenarios

Based on the system’s capabilities, new scenarios for

synchronous and asynchronous distributed learning evolve.

Actually the system is designed for and used in various

learning scenarios.

(i) Synchronous distributed learning scenario

Teacher and students are connected by LAN or WLAN.

All participants establish mutual connections via web server.

The teacher can than send his PC presentations and

applications to the students PCs. Students (outside or inside

the lecture room) can participate active and/or passive by

real-time audio/video with latencies (in LANs) well below

50 ms. Since all participants can send their presentations or

applications via WLAN to a beamer in a conference room,

this can be used as a “wireless” connected beamer which

can present full video formats.

All participants can also initiate co-operations in small

groups via full video conferencing. Within the peer to peer

network each student can send, receive and work on any PC-

applications for collaboration.

Additionally there is also a live streaming option. Any

participant is able to send one audio/video stream to a

preconfigured windows media streaming server via push

distribution. This client works like an A/V gateway. All

H.264 video conferencing video streams will be mixed to

one video stream and then live-transcoded into windows

media format. So additionally also passive participants can

join the videoconference, without overloading the

bandwidth between the peer to peer videoconferencing

groups. The streaming server adapts the available

connection bandwidth of the passive viewers by using multi-

bit rate transcoding profile.

Fig. 1. Video Conferencing Live Streaming

Thus all viewers get the best optimization for there

appropriate connection bandwidth.

(ii) Asynchronous distributed learning scenario

Each station can record all sessions. The recordings can

be stored locally or made net-wide accessible a by

converting it e.g. into a MS streaming format and uploaded

to an e-learning platform. Lecture room presentation or

distributed group work are thus ready to be played back

anywhere at any time by streaming video.

III. MOBILITY AWARE SOURCE SPECIFIC MULTICAST

COMMUNICATION

A. The Mobile Multicast Source Problem

Mobility today must be seen as one of the major driving

forces for multimedia data transmission. Cellular phones

and portable paddles we expect to carry individual Internet

addresses soon, as available from IPv6 address space, and to

operate mobility supporting Internet protocols as the

recently released MIPv6 [7]. Multimedia applications s. a.

our video conferencing system will request seamless support

for mobile group conferencing, thereby occurring as

simultaneous multicast sender and receiver to the Internet

infrastructure.

Source mobility presents a severe problem for multicast

packet distribution. Even though multicast routing itself

supports dynamic reconfiguration, as members may join and

leave ongoing group communication over time, multicast

group membership management and routing procedures are

intricate and too slow to function smoothly for mobile users.

In addition multicast imposes a special focus on source

addresses. Applications commonly identify contributing

streams through source addresses, which must not change

during sessions, and routing paths in most protocols are

chosen from destination to source.

Routing overheads in Any Source Multicast (ASM) [12]

up until now have hindered the widespread availability of

multicast services. It is currently expected that the

simplified, interdomain-transparent group communication

scheme of Source Specific Multicast (SSM) [11] will offer

the basis for widely available group communication support

at the network layer.

Source addresses in source specific multicast are

requested to be known prior to group subscription. They

need to be shared by the entire group of conference

members and thus are to be provided by the signaling

scheme for group initiation. The latter can be achieved by

appropriate SIP [9] negotiations. Additionally, source

addresses carry the dual meaning of client–subscribed

source–group identifiers at the one hand, and routing

location information at the other hand. Both semantics need

to be followed by receivers and intermediate routers.

The ‘lightweight’ SSM approach to group communication

can thus be considered as a highly appropriate Internet

solution for multimedia conferencing, but enforces

extensions to session initiation and mobility management.

B. Session Initiation for Source Specific Group Conferencing

Session initiation for conferencing applications

commonly is negotiated by the session initiation protocol

SIP [9]. Within its invite message SIP reports on

contributing node parameters and media session specific

characteristics via an SDP data set. SIP accounts for

multicast group conferencing by allowing a multicast

distribution of its signaling via an maddr header field.

However, SIP core procedures are bound to Any Source

Multicast communication, which significantly simplifies the

coordination of primarily unknown members of the

distributed conferencing system.

To enable group communication by Source Specific

Multicast, SIP dialogs need alteration in the following way:

A new conferencing member willing to join a previously

established group conference invites any party and receives

acknowledgement including multicast session descriptions

via unicast. The invited party then has to repeat the

acknowledgement to a previously established SSM signaling

domain, in order to trigger an active source subscription of

all previous group members to the newly established caller.

All additional group members subsequently will advertise

their session affiliation, while the initially called party will

signal a turnover of its newly established SIP signaling

channel to SSM multicast. As soon as the new conferencing

member has completed subscription to SIP signaling and

media session groups for all conference party's addresses, a

Source Specific Multicast group conference is fully

established among peer-2-peer members in the absence of

any coordinating instance.

C. Tree Morphing – Introducing Multicast Routing Trees Adaptive to Mobility

In the present section we will briefly introduce our new

concept of multicast routing, adaptive to source mobility. A

mobile multicast source (MS) away from home will transmit

unencapsulated data to a group using its Home Address

(HoA) on the application layer and its current Care-of-

Address (CoA) on the Internet layer, just as unicast packets

are transmitted by MIPv6. Likewise data packets will carry

a mobility destination option header to pass HoA as source

identifier to the application layer at the receiver side. In

extension to unicast routing, though, the entire Internet

layer, i.e. routers included, will be

R

R R

D2D1

R

R

R RpDR nDR

MS MS

Fig. 2. Elongation of the Root of the Delivery Tree

aware of the permanent HoA. Maintaining address pairs in

router states like in binding caches will enable all nodes to

simultaneously identify (HoA,G) – based group membership

and (CoA,G) – based tree topology.

When moving to a new point of attachment, the MS will

alter its address from previous CoA (pCoA) to new CoA

(nCoA) and eventually change from its previous Designated

multicast Router (pDR) to a next Designated Router (nDR).

Subsequent to handover it will immediately continue to

deliver data along an extension of its previous source tree.

Delivery is done by elongating the root of the previous tree

from pDR to nDR (s.Fig. 2). This extension is achieved

through a state update message, carried in a Hop-by-Hop

option header and sent to the multicast destination address

using source routing through pDR. All routers along the

path, located at root elongation or previous delivery tree,

thereby will learn MS’s new CoA and implement

appropriate forwarding states.

Routers on this extended tree will use RPF checks to

discover potential short cuts. Registering nCoA as source

address, those routers, which receive the state update via the

topologically incorrect interface, will submit a join in the

direction of a new shortest path tree and prune the old tree

membership, as soon as data arrives. All other routers will

simply overwrite their (pCoA,G) state with

(nCoA,G).Thereby all parts of the previous delivery tree,

which coincide with the new shortest path tree, are re-used.

Only branches of the new shortest path tree, which have not

previously been established, need to be constructed. In this

way the previous shortest path tree will be morphed into a

next shortest path tree as shown in figure 3 and 4.

R

R R

D2D1

R

R

R

R RpDR nDR

MS MS

Fig. 3. Intermediate Morphing State

R

R R

D2D1

R

R

R RpDR nDR

MS MS

Fig. 4. Final Morphing State

Note that this algorithm does not require data

encapsulation at any stage. It is not built upon a specific

multicast routing protocol, but will require the following

functional mechanisms compliant with current protocols

such as PIM-SM [13]:

• Outgoing router interfaces need to maintain (S,G) states to denote their partition in the distribution tree. These states will be extended to include the Home Address identifier (S, G, HoA).

• Routers need the ability to explicitly join an (S,G) state.

• Routers need the ability to explicitly prune an (S,G) state. Alternatively, but with lower efficiency, routing states may time out.

• Finally, the computation of standard Reverse Path Forwarding (RPF) check is used.

For the details of signaling and routing protocol

extensions under SSM mobility we refer the reader to [14].

D. Performance Evaluation

Mobility initiated handovers may in general lead to

packet loss and delay. The tree morphing multicast routing

scheme will not produce any packet loss in addition to

mobile IP handovers, as can be easily concluded from

primary packet forwarding relying on unicast source routes.

For a first evaluation measure we will subsequently

concentrate on handover initiated packet delay as a result

from initially suboptimal delivery trees. Based on real-world

Internet topologies we simulate the packet distribution and

compare our results to the bi-directional tunneling approach

[7], which currently is the only stable mobility solution for

SSM source mobility.

To judge on performance quality of the tree morphing

(TM) scheme, we now analyze its delay effects within

realistic Internet topologies. We per-formed a stochastic

discrete event simulation based on the network simulator

platform OMNeT++ 3.1 [10] and several real–world

topologies of different dimensions. The selection of network

data in our simulation must be considered critical, as key

characteristics of multicast routing only make an impact in

large networks, and as topological setup fixes a dominant

part of the degrees of freedom in routing simulations.

We chose the ATT core network [15] as a large (154

nodes), densely meshed single provider example. For inter–

provider data we extracted sub-samples of varying sizes

from the ”SCAN + Lucent” map [19, 20], the result of two

extensive Internet mapping projects containing 284.805

network nodes connected by 449.246 links. Sample sizes,

154 and 15.400 nodes, vary by two orders of magnitude.

The delay excess relative to optimal routes has been

calculated as characteristic performance measure under the

assumption of homogeneous link delays. Extreme values,

i.e. maximal delays at initial elongation phase and minimal

after convergence, were evaluated for tree morphing (TM)

as functions of the distance from pDR to nDR. In detail,

designated routers within a given topology were randomly

chosen according to their predefined distances. For each pair

of edge routers at the mobile source a uniformly distributed

set of 20 receivers was established and delay values were

taken from average reception time. Sampling of source

positions was repeated 20 times for each parameter set in

order to better explore the large phase space. Comparisons

are drawn with bi-directional tunneling (BT), which does

not depend on designated router distances, but on home

agent (HA) position. The delay excess in BT as function of

HA position does not converge to a characteristic value, but

rather admits a broad distribution. The latter has been

derived from scattering HA positions uniformly within the

sample networks.

The results of our simulations are displayed in figure 5

and 6. pDR to nDR distances were chosen between 2 and

10, except for the ATT network, which exhibits a maximal

edge router separation of 5.

Fig. 5. Internet 15.400 N odes

Fig. 6. ATT Core Network

Error bars indicate the standard deviation of initial TM

delay excess, as calculated from events differing in location

of the mobile source. Plotted lines indicate the linear

regression curves derived from this result set. Delay excess

distributions for scattered HAs in BT are laid underneath

TM curves in grey dots.

It can be observed that initially maximal delays of the tree

morphing scheme tend to remain below the average of

permanent BT packet retardation. Convergence of the TM

then will lead to (relatively) undelayed packet delivery,

which is never met in BT. Little dependence on network size

becomes visible for TM — relative delays more strongly

change with topologic characteristics. In a densely meshed

provider network such as the ATT core, packet transitions

are rapid and therefore initial delays from tree elongation

account more dominantly for our relative measure. In the

contrary it is interesting to note that delays from BT admit a

systematic dependence on network size: BT average delay

excess increases from 45 % in the small ATT network to

about 120 % at sample size 15.400. From these observations

it can be concluded that bi-directional tunneling attains

appropriate performance for small communities within a

densely meshed core network, but becomes infeasible in

large inter-provider domains. The tree morphing even in its

initially weakest phase exhibits fairly uniform performance,

no matter how large the underlying network is.

IV. CONCLUSION AND OUTLOOK

In this paper we presented a distributed communication

conferencing software and some of its applications. The

applications include an easy-to-use scheme for distributed

presenting, recording and streaming of multimedia content.

The video conferencing module is based on a H.264/MPEG-

4AVC software implementation. In addition to this we use

also some costumer tailored wavelet-based codecs.

Concordantly we presented an approach to solve the mobile

source problem in SSM routing. This novel scheme of

morphing a previous distribution tree into a new shortest

path tree operates based on common multicast routing

protocols with simple algorithmic extensions. After a

handover it allows for immediate data transmission and

strictly avoids tunneling. All procedures are robust and of

rapid convergence. First performance simulations indicate

an overall low initial delay of the tree morphing scheme,

outperforming the conventional bi-directional tunneling

approach.

In future work we will optimize the communication

system for more inhomogeneous networks and quantify

further characteristic measures of the scheme by

simulations.

REFERENCES

[1] ITU-T Recommendation H.323: “Infrastructure of audio-visual services – Systems and terminal equipment for audio-visual services:

Packet-based multimedia communications systems”, Draft Version 4, 2000.

[2] The daViKo homepage, http://www.daViKo.com, 2005.

[3] J. Ostermann, J. Bormans, P. List, D. Marpe, M. Narroschke, F. Pereira, T. Stockhammer, and T. Wedi : Video Coding with H.264 / AVC: Tools, Performance, and Complexity , IEEE Circuits and

Systems Magazine, vol. 4, no. 1, pp. 7-28, April 2004. [4] H. L. Cycon, M. Palkow, T. C. Schmidt, M. Wählisch, and D. Marpe:

“A Fast Wavelet-Based Video Codec and Its Application in an IP

Version 6-Ready Server less Videoconferencing System.”, International Journal of Wavelets, Multiresolution and Information Processing (IJWMP), vol. 2, no. 2, pp. 165-171, June 2004

[5] The Speex Project homepage, http://www.speex.org, 2005 [6] T. C. Schmidt, M. Wählisch, H. L. Cycon, and M. Palkow, “Global

serverless videoconferencing over IP”, Future Generation Computer

Systems, vol. 19, no. 2, pp. 219–227, February 2003. [7] D.B. Johnson, C. Perkins and J. Arkko: “Mobility Support in IPv6”,

IETF RFC 3775, June 2004.

[8] http://www.javvin.com/protocolRUDP.html [9] J. Rosenberg, H. Schulzrinne, et al, "SIP: Session Initiation Protocol",

IETF, RFC 3261, Juni 2002.

[10] A. Varga et al., “The OMNeT++ discrete event simulation system,” http://www.omnetpp.org, 2005.

[11] S. Bhattacharyya, “An Overview of Source-Specific Multicast

(SSM),” IETF, RFC 3569, July 2003. [12] S. E. Deering, “Host Extensions for IP Multicasting,” IETF, RFC

1112, Aug. 1989.

[13] B. Fenner, M. Handley, H. Holbrook, and I. Kouvelas, “Protocol Independent Multicast - Sparse Mode (PIM-SM): Protocol Specification (Revised),” IETF, Internet Draft – work in progress 11, October 2004.

[14] T. C. Schmidt and M. Wählisch, “A First Performance Analysis of the Tree Morphing Approach to Source Mobility in Source Specific

Multicast Routing”, in Proceedings of the IEEE ICN'06, IEEE Press,

2006. [15] O. Heckmann, M. Piringer, J. Schmitt, and R. Steinmetz,“On Realistic

Network Topologies for Simulation,” in MoMeTools ’03:

Proceedings of the ACM SIGCOMM workshop on Models, methods and tools for reproducible network research. New York, NY, USA: ACM Press, August 2003, pp. 28–32.

[16] J. Rosenberg, J. Weinberger, C. Huitema, R. Mahy: “STUN - Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)”, IETF, RFC 3489, March 2003.

[17] T. C. Schmidt and M. Wählisch, “Multicast Mobility in MIPv6: Problem Statement”, Internet Draft – work in progress, October 2005.

[18] I. Romdhani, M. Kellil, H.-Y. Lach, A. Bouabdallah, and H.

Bettahar, “IP Mobile Multicast: Challenges and Solutions,” IEEE Comm. Surveys & Tutorials, vol. 6, no. 1, pp. 18–41, 2004.

[19] “SCAN project: Internet Maps. SCAN+Lucent

map,”http://www.isi.edu/scan/mercator/maps.html, 2005. [20] R. Govindan and H. Tangmunarunkit, “Heuristics for internet map

discovery,” in Proceedings IEEE INFOCOM 2000, vol. 3. Tel Aviv,

Israel: IEEE Computer Society, March 2000, pp. 1371–1380.

Hans L. Cycon is currently teaching mathematics and signal processing at

FHTW Berlin, University of Applied Sciences. He received his diploma in physics in 1975 and his PhD in mathematics in 1979 and his habilitation in 1984 from the Technical University Berlin, Germany. His publications

fields are mathematical physics and signal processing i.e. image coding. Hans L. Cycon is leading several projects in developing wavelet based still image and video compression codecs. He is member of the German

delegation of the ITU/ISO standardization committee for JPEG 2000 still image standard.

Thomas C. Schmidt is teacher of Information Engineering at the HAW Hamburg and project manager at FHTW Berlin, where he was head of the computer centre for many years.

He studied mathematics and physics at Freie Universität Berlin and University of Maryland, USA. In 1993 he received his PhD in mathematical physics for a work in many particle theory of quantum

mechanics done at the theory group of the Hahn-Meitner-Institut Berlin. Since the late 1980s he has been involved in many computing projects, especially focusing on simulation and parallel programming, distributed

information systems and visualisation. His current fields of interest lie in the areas of mobile and multimedia networking and hypermedia information processing, where he has continuously conducted numerous

projects on national and international level.

Matthias Wählisch is a member of the networking group of the computer

centre of FHTW Berlin. He is studying mathematics and computer science at Freie Universität Berlin. His major fields of interest lie in networking protocols, where he looks back on seven years of professional experience in

project work and publication.

Mark Palkow presently is the Managing Director and Chief Developer at

the daViKo Gesellschaft für digitale audiovisuelle Kommunikation mbH that he founded in 2000. He received his diploma in communication engineering from the Fachhochschule Telekom Berlin in 1996. Since then

he has worked on several research projects at FHTW Berlin, the Old Dominion University Norfolk and the Heinrich Hertz Institut Berlin.

Henrik Regensburg is a member of the developer group of the “competence center media and networks” of FHTW Berlin. His major fields of interest lie in distributed video applications and coding,

networking and a/v-content authoring. He received his diploma in applied computer science from the University of Applied Sciences FHTW-Berlin in 2002. Since then he has worked on research projects at FHTW Berlin and

several freelance projects in commerce, all concerned with video conference technology and e-learning.