AR CHI T E CT S OF AN I NT E R NE T WOR L D

Connecting Mobile
Videophones to Wireline
Broadband Video Services
T E C H N O L O G Y WH I T E P A P E R
With the introduction of both the ISDN and
intelligent terminals complying with
Recommendation H.320, video, in addition to
the voice service in user terminals, has been a
technical and commercial reality since the
beginning of the 1990s. This is why
videoconference terminals connected in N x 64
kbit/s mode have been so successful in
companies. To reach the consumer market and
offer audio/video services over 64 kbit/s
circuits, we had to wait for processor power to
support more efficient video compression. In the
second half of the 1990s, the H.324 standard
laid down the foundations for future consumer
market videotelephony in the circuit environment.
Although the standard has been stabilized for
more than five years, we have had to wait for
UMTS and its new high bit rate circuit
capabilities for terminal manufacturers to offer
videotelephony services. The interconnection of
this UMTS service with other video services
already available in the IP environment should
significantly increase the chances of the service
being a success with the users.
Mobile operators can increase traffic by connecting 3G videophones
to fixed network video services.
A. Bultinck, A. Couturier, M. Tadault
CONNECTING MOBILE
VIDEOPHONES TO WIRELINE
BROADBAND VIDEO SERVICES
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Introduction
The deployment of the first Universal Mobile
Telecommunications System (UMTS) networks marks the
return of the dream of mass market videotelephony. This
service, based on the 3G-324m standard, lets users see
and hear the person at the other end of the line. Although
UMTS technology is primarily known for high speed
packet transport, the videotelephony service uses
64 kbit/s circuit-based transport. In fixed networks, the
proliferation of high speed Internet Protocol (IP) access
networks, whether using Digital Subscriber Line (xDSL)
or cable, means that we can expect consumer
videotelephony to be implemented over IP using protocols
like H.323 and the Internet Engineering task Force
(IETF) Session Initiation Protocol (SIP). Even the
business world, which has deployed Integrated Services
Digital Network (ISDN) videotelephony equipment
(Recommendation H.320), will switch over to IP.
Unlike the first Global System for Mobile
communication (GSM) users, who could make or receive
telephone calls on the millions of fixed line terminals
already deployed, the first UMTS users that subscribe to
videotelephony have no option other than to make or
receive video calls to or from other
UMTS users with the same video
facilities. Since initially this will be
only a small number of users, the
service is not particularly attractive
and generates little revenue. It is
therefore logical to offer a gateway
that interconnects UMTS
videotelephony with IP
videotelephony. Given the millions
of high speed Internet access ports
already deployed, the number of
users is much greater and the
service more attractive, with the
result that traffic will grow.
A derivative of this
interconnection gateway allows
UMTS videotelephony users to view video sequences from
IP streaming servers. Known as circuit-switched
streaming, as opposed to packet-based streaming, its
value lies in the quality of service it can provide for long
video sequences. This gateway can adapt IP streaming
sources (IP being the most common form for a video
source over the Internet) to a videotelephony circuit.
Video Compression Standards
Digital video streaming is bandwidth intensive and thus
difficult to implement without compressing the video
information sufficiently to make it compatible with the
network bandwidth. Video compression/decompression
algorithms are CPU-intensive as they have to analyze, in
real-time, video bitmaps with streaming frequencies as high
as 20 images per second (30 images per second for the
highest qualities). The International Telecommunications
Union (ITU) and International Standards Organization /
Motion Picture Experts Group (ISO/MPEG) have defined
various video compression standards for videoconferencing
over ISDN (H.320 standards) and digital video storage on a
PC (MPEG-2 standards).
Briefly, these standards are:
Sub-quarter CIF
Quarter CIF
CIF
4CIF
16CIF
CIF format
Video coders operate on successions of bitmaps which currently comply with the pivotal Common Intermediate Format (CIF) or its
multiples and submultiples (see table below). These bitmaps contain pixels encoded using eight bits representing luminance
information (gray levels) and chrominance information (levels of blue and red coloration). Since the human eye is much more
sensitive to luminance than chrominance, the CIF format uses it to define a luminance pixel/chrominance pixel concentration ratio
of 4: an 8 X 8 pixel luminance bitmap region contains 4 X 4 red chrominance pixels and 4 X 4 blue chrominance pixels.
128
176
352
704
1408
96
144
288
576
1152
64
88
176
352
704
48
72
144
288
576
Image format
Number of luminance
pixels per line
Number of lines
for luminance
Number of chrominance
pixels per line
Number of lines
for chrominance
Tab. 1 CIF format and its derivatives
H.261: This specification
covers a reference
coder/decoder (codec) for
ISDN videoconferencing
compatible with the H.320
standard. An H.261 coder
requires a bandwidth of more
than 64 kbit/s to encode a
video channel.
H.263: Defines a more
powerful video coder that
enables video/audio services to
be offered over a single
64 kbit/s channel, the main
objective being point-to-point
videotelephony.
MPEG-4: This ambitious
multimedia communication
standard includes definition of
scene, audio encoding,
synthesis and natural video
encoding. With an eye on
compatibility, part 2 defines a
pure video profile that
provides some compatibility
with H.263.
H.264: Defined jointly by ITU
and MPEG, this has the same
scope as H.263 in terms of
bandwidth requirements. It
should realize bandwidth savings of around 33%
over H.263 at the cost of a two- to three-fold
increase in CPU power.
Video Compression Principles
Tables 1 and 2 briefly outline the major video
compression principles.
H.320 Videotelephony
The H.320 standard defined by the ITU in 1989
essentially popularized N 64 kbit/s ISDN videoconference
systems with business users. Depending on its capacity,
such a system sets up narrowband ISDN channels with its
remote partners to support the different voice, video and
data channels. The main characteristics are:
Call handling is provided by Digital Subscriber
Signaling System 1 (DSS1), as defined in
Recommendation Q.931.
Multimedia channel multiplexing and the
multimedia control protocol are defined in the
H.221 and H.242 standards.
H.324 Videotelephony
The H.324 standard defined by the ITU in 1996
proposes a set of protocols for offering multimedia
services in a 64 kbit/s circuit-based environment. It
currently applies to Public Switched Telephone Network
(PSTN) terminals, 64 kbit/s ISDN terminals (H.324I) and
UMTS mobile terminals (3G-324m). The standard has
not had great commercial success to date in either the
PSTN or ISDN, and few terminals are available. However,
its application in the mobile environment was of
immediate interest to UMTS operators who saw in it a
new 3G service different from 2G. Mobile terminal
manufacturers quickly introduced this service into their
new terminals.
The H.324 standard is structured as follows.
Call handling
Call control protocol is the network access protocol
implemented by the terminal to set up the 64 kbit/s
circuit supporting the different multimedia channels. It
clearly depends on the access technology: DSS1 (Q.931)
for H.324I terminals and Third Generation Partnership
Project (3GPP) TS 24.008 for 3G-324m terminals.
64 kbit/s circuit multiplexing
This is the multiplexing structure definition above the
64 kbit/s circuit that makes it possible to transport each
multimedia flow (voice flow and video flow). The protocol
is common to all H.324 terminals (PSTN, ISDN or mobile).
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Temporal and spatial compression
Video coders structure the input video bitmap into blocks (8 pixels X 8 pixels), macro blocks (16 pixels X 16 pixels) and block
groups (fixed number of macro blocks, depending on the bitmap image format). One area of the bitmap corresponding to a
luminance macro block therefore contains four luminance blocks and one chrominance block for each of the two colors.
Temporal compression
To minimize the information transmitted for a video frame, the coder looks to see whether or not the pattern can be deduced
from an adjacent macro block in the previous frame for each of the frames luminance macro blocks (and its two related
chrominance blocks). If it succeeds, instead of sending pixels, it sends a movement vector which indicates the movement of the
macro block in relation to the previous frame. If necessary, it supplements this vector with residual coding information (coding of
the differences between the macro block of the frame and its recognized image in the previous frame) if the macro block pattern
has changed slightly.
A frame containing temporally compressed blocks is called P (Predictive frame), while one containing spatial compression only
is called I (Intracoded frame). The coder regularly generates I frames for resynchronization purposes; errors in P frames are
propagated from frame to frame.
Spatial compression
Spatial compression involves a series of operations that significantly reduces the information transmitted for each block:
Discrete Cosine Transform (DCT): The coder translates the block matrix into a matrix that reduces redundancy; the more
uniform the block, the less information it will contain.
Quantization: The coder assigns a reduction factor to each 8-bit element of the matrix to reduce the number of useful bits
transmitted. This operation, unlike the others, involves some loss of information.
Variable Length Coding (VLC): The coder reduces the number of bits sent by assigning bit codes to each byte occurrence;
the most frequent bytes are expressed using the shortest code.
Synchronization in the frame
The effectiveness of spatial coding which breaks the byte structure of the initial video frame is directly dependent on
synchronization within the frame itself. The transmission of an errored bit will, more often than not, disrupt decoding of the
entire frame! For this reason, the I and P frames are divided by the coder into Groups of Blocks (GoB), with each group being
preceded by a synchronization pattern.
Tab. 2 Video compression principles
It is the H.223 standard, which is based on High level Data
Link Control (HDLC) frame structures. HDLC flags are
sent between the H.223 frames. Like the Asynchronous
Transfer Mode (ATM), the H.223 protocol is itself divided
into two levels:
Multiplexing level, which defines the HDLC
structure.
Adaptation Layer (AL), which defines the
adaptation protocols used to transport the
application Protocol Data Units (PDU).
Three adaptation layers are defined: AL1 for data
transport, AL2 for voice transport and AL3 for video
transport.
Media control
The protocol controlling the media is defined in
Recommendation H.245. It allows the two end points to
negotiate the logical channels supporting the different
media. The main functions of H.245 are used to:
Describe the transmit and receive mode terminal
capabilities: types of media supported, number of
parallel media, etc.
Describe the multiplexing structures supported in
the transmit and receive modes.
Open/close the logical channels to transport the
media from end to end.
Coders
Since the maximum available bandwidth is 64 kbit/s, the
traditional G.711 voice coder is clearly of no interest.
H.324 recommends the use of packet-based voice coders,
such as G.723.1 (G.723.1 offers two coding options:
5.1 kbit/s or 6.1 kbit/s), as they consume little bandwidth.
The Adaptive Multi-Rate (AMR) codec used in the 3GPP
voice circuit service has a low bandwidth requirement and
is a good candidate for 3G terminals (as already
implemented in 3G terminals).
The video coder recommended by H.324 is the H.263 coder.
Note that MPEG-4, in its pure video profile (MPEG-4,
part 2), proposes similar algorithms to those in H.263;
this alternative is widely implemented in Asia for mobile
terminals.
H.323 Videotelephony
H.323 and SIP together comprise the reference
standards for multimedia exchange in the Internet
environment.
H.323 encompasses a set of subsidiary standards:
H.225 for call control, H.245 for media control, and Real-
time Transport Protocol / User Datagram Protocol /
Internet Protocol (RTP/UDP/IP) for multiplexing, etc.
The standard was defined by the ITU in parallel with
H.324; it was also published at the beginning of 1996.
Because it came out at the same time, many of the
technical choices are equivalent: H.245 media control
protocol, recommended H.263 video coders, etc.
The H.323 standard is designed to be used in different
environments, ranging from small private networks to
public networks. To meet this requirement, it proposes
several architectures:
Point-to-point architecture in which users
communicate directly, with no intermediate
network element (Direct Routed Mode). H.225
call control messages and H.245 media
negotiation messages are exchanged directly
between the users.
Architecture more like a traditional
telecommunication network in which the network
deals with call handling (setting up the bearer)
and the media channels are negotiated between
the network end points (Semi Routed Mode).
H.225 call control messages are exchanged
through a gatekeeper, while H.245 media
negotiation messages are directly exchanged
between users.
Architecture corresponding more to the next
generation network evolution in which the
network participates in both call control and
media negotiation (Full Routed Mode). H.225 and
H.245 messages are exchanged through a
gatekeeper.
A semi-routed or full routed H.323 network comprises
the following elements:
Gatekeepers to manage IP users: As in a
conventional class 5 switch, the users are
provisioned in a gatekeeper and identified by a
public address of either the E.164 or a Universal
Resource Locator (URL) type. To resolve the
public address / IP transport address translation
and to assure subscriber access / authentication
security, the call handling phase is preceded by a
subscriber registration phase in which the H.323
terminal provides its public address and its IP
transport address, and runs an authentication
procedure with the gatekeeper.
H.323 terminals.
H.323 access gateways for communication with
other types of multimedia network: 3G-324m
gateway to H.323 for communicating with 3G
Public Land Mobile Networks (PLMN), H.320
gateway to H.323 for communicating with ISDN
networks, etc.
ITU defined the H.323 environment to facilitate
interconnection with multimedia environments on H.324-
compliant and, to a lesser extent, H.320-compliant circuits:
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H.323 call handling protocol defined in H.225 is
closely based on Q.931.
Media control protocol is H.245, as in the H.324
environment.
H.323 supplementary services, defined in H.450.x,
reprise the supplementary services defined in the
Q.73x recommendations: number presentation, call
forwarding, etc.
Extensions of the Mobile Peer-to-Peer 3G-
324m Videotelephony Service
To broaden the coverage and to pave the way for the
future success of the 3G.324m videotelephony service,
mobile operators are looking to provide interworking with
other audio and video services.
3G-324m videotelephony interconnection to IP H.323
networks
One of the most promising avenues is the interworking
of mobile videophones with PCs equipped for high speed
Internet access. PCs fitted with H.323 client software
(such as Netmeeting) can participate in videophone
conversations with mobile videophones by registering
with the mobile operators dedicated gatekeeper. The
architecture for this service is shown in Figure 1.
In addition to the mobile network elements, the
architecture comprises the following elements.
3G-324m to H.323 gateway
This gateway can be connected to the mobile network
via a NetworkNetwork Interface (NNI) using the ISDN
User Part (ISUP) protocol, or to a UserNetwork
Interface (UNI) using the DSS1/Q931 protocol, as shown
in Figure 2. At the user plane level, the gateway is
responsible for:
Converting H.223 multiplexing into H.225
multiplexing: The recommended multiplexing in
H.225 is the natural multiplexing offered by the IP
network; each medium is transported in a different
RTP/UDP/IP microflow.
Voice transcoding: The AMR codec, which is the
only one implemented in mobile terminals, is not
yet popular in non-mobile environments. Current
H.323 applications on PCs generally use G.711 or
G.723.1. The gateway therefore has to provide two
conversions: AMR/G.711 and AMR/G.723.1.
Video transcoding: Depending on the mobile
terminal manufacturer, two standards (MPEG-4
and/or H.263) may be implemented. In the H.323
environment, only H.263 and H.261 are currently
available. Therefore, should the mobile only
implement MPEG-4, the gateway must offer
MPEG-4/H.263 or MPEG-4/H.261 conversion.
At the control plane level, the gateway:
Translates circuit-based call handling (ISUP or
DSS1) into H.225 on Transmission Control
Protocol / Internet Protocol (TCP/IP).
Relays the H.245 protocol, carried on a TCP/IP
connection in the H.323 environment and on
channel 0 of the H.223 multiplex
in the H.324m environment.
H.323 gatekeeper
The H.323 gatekeeper is relatively
standard. It handles the registration
of H.323 IP subscribers and routes
their calls between the two
environments from the gateway to
the H.323 applications on the PC.
Alcatel solutions
Alcatel can offer mobile
operators a number of solutions to
increase the penetration of video
services in their 3G networks.
These solutions, which are already
being sold to some 3G operators,
are based on the Alcatel 5020 IPT
gatekeeper and on a third-party
gateway.
This basic service allows 3G
phones to have peer-to-peer
videotelephone communication with
PCs connected to broadband.
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H.324m/H.323
interconnection
service
H.323 terminal
Firewall
Internet
Intranet
Gatekeeper
H.324m/H.323
gateway
H.324m
terminal
RNC Node B VMSC MSC
Fig. 1 Network architecture for the H.324m/H.323 interconnection service
MSC: Mobile Switching Center RNC: Radio Network Controller VMSC: Visited MSC
Alcatel is currently studying the technology and
proposing ways in which it can evolve to track this
emerging market:
mix mobile/fixed multimedia conference services;
peer-to-peer mobile communication with ISDN
H.320 terminals;
evolution to IP user with SIP environment.
Circuit-switched streaming
Another area for investigation is the diffusion of the
existing video service in the IP environment (live
services like TV channel distribution, or other services
like movies on demand) using the 3G-324m circuit-
switched service. These media streaming services are
traditionally based, for the user plane, on the continuous
transmission of video and audio frames in RTP/UDP/IP
packets. If this type of service is to be acceptable to
users (i.e. they will be willing to pay for them), they
must comply with stringent Quality of Service (QoS)
constraints, especially in terms of bandwidth availability
and jitter tolerance. These constraints are expected to be
met in 3G networks when the packet-switched
streaming class becomes available. It will then be
sufficient for the terminal to implement the IETF
streaming protocol suite to interface with the streaming
servers already available on the market (Real Network,
Packet Video, etc):
Through the HTTP-like Real-Time Streaming
Protocol (RTSP) / TCP / IP to pilot the server (set
up to select the media stream, play to launch the
media downstream emission, tear-down to stop the
downstream emission).
Through RTP/UDP/IP to receive the different video
and voice media streams.
Until this target architecture is available, there is a
window of opportunity during which it will be possible to
offer gateway facilities for converting existing streaming
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Q.931 translation
H.245
translation/conversion
RTP/H.223 channel
translation
Media conversion
H.223
multiplex
Q.931
Q.921
Ch0
Chx
RAS control
RAS
control
port
Call
control
port
Media
control
port
RTP/RTCP
ports
IP stack
ernet eee Ethhe
B-channels
D-channel
S0/T0
H.223
multiplex
H.223
multiplex
Fig. 2 Functional architecture of an H.324m/H.323 gateway connected in UNI mode
protocols used in the streaming server to H.324m
protocols used in 3G videophone terminals. The 3G
circuit-switched domain already provides the necessary
QoS; it even offers conversational class, which is better
than required. The basic idea is to provide a gateway and
a dedicated MSC that will do the job, that is:
Deduce from the mobile call control / media
control (H.245) commands the set of RTSP
commands that have to be sent to the streaming
server.
Translate the RTP/IP media flow coming from the
video streaming server onto H.223 channels (same
as for H.323 H.324m media translation).
Similarly to the video H.324m H.323 gateway, the
H.324m streaming gateway can implement some codec
conversion if the servers and terminals are incompatible.
Alcatel is proposing a set of H.324m streaming gateway
solutions to its customers which can be combined with
H.324m H.323 gateway solutions.
Conclusion
The gateways interconnecting the UMTS (3G-324m), IP
(H.323/SIP) and ISDN (H.320) videotelephony systems
are helping to make videotelephony much more popular
by breaking down the barriers between networks. It is in
the interests of both fixed and mobile telecoms operators
to deploy this type of technology because their
videotelephony traffic will be considerably increased by
the larger number of potential users.
A derivative of this type of gateway lets UMTS
operators provide circuit-switched video streaming. The
quality of service offered by this medium, enables them to
offer long video streams as opposed to the video clips
available today in 2G (download and further play
model). Consequently, we can envisage real-time
streaming of information-providing television channels.
These video services have been made possible on
mobile terminals by the relentless increases in processor
power and miniaturization. With this trend continuing, the
latest H.264 video coder could herald video services of
still better quality, even on mobile terminals.
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Alain Bultinck is a Mobile System Engineer in charge
of defining the video gateway architecture in the Chief
Technology Office team of the Mobile Communications
Group, Vlizy, France. (Alain.Bultinck@alcatel.fr)
Alban Couturier is Product Manager for the video
gateway solution in the Mobile Network Division of the
Mobile Communications Group, Vlizy, France.
(Alban.Couturier@alcatel.fr)
Michael Tadault is Mobile Internet Product Manager
in the Mobile Networks Division of the Mobile
Communications Group, Vlizy, France.
(Michel.Tadault@alcatel.fr)
Abbreviations
3GPP Third Generation Partnership Project
AL Adaptation Layer
ATM Asynchronous Transfer Mode
CIF Common Intermediate Format
CPU Central Processing Unit
DCT Discrete Cosine Transform
DSS1 Digital Subscriber Signaling System 1
GoB Group of Blocks
GSM Global System for Mobile Communication
HDLC High level Data Link Control
HTTP HyperText Transfer Protocol
IETF Internet Engineering task Force
IP Internet Protocol
ISDN Integrated Services Digital Network
ISO International Standards Organization
ISUP ISDN User Part
ITU International Telecommunications Union
MPEG Motion Picture Experts Group
MSC Mobile Switching Center
NGN Next Generation Networks
NNI Network to Network Interface
PDU Protocol Data Unit
PLMN Public Land Mobile Networks
PSTN Public Switched Telephone Network
QoS Quality of Service
RNC Radio Network Controller
RTCP RTP Control Protocol
RTP Real-time Transport Protocol
RTSP Real-Time Streaming Protocol
SIP Session Initiation Protocol
TCP Transmission Control Protocol
UDP User Datagram Protocol
UMTS Universal Mobile Telecommunication System
UNI UserNetwork Interface
URL Universal Resource Locator
VLC Variable Length Coding
VMSC Visiting Mobile Switching Center
xDSL Digital Subscriber Line
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