MOBILE VIDEO TELEPHONY

3RD Generation Technology

What is 3G and why 3G?
3G is the third generation development in
mobile telephony technology. It succeeded the
two earlier technologies.i.e. 1st & 2nd generation.

The International Telecommunication Union

(ITU-T) defined the third generation technology
to increase the bandwidth (data carrying
capacity), support more diverse applications e.g.
mobile video telephony, and to facilitate
revenue growth for telecom operators.
Mobile Video Telephony?
Mobile video telephony is the mobile visual
communication with video sequences and
speech transmitted simultaneously over the
telecommunication network.

It is a real-time system of the conversational

type in which the playback of continuous
media such as audio and video occur in an
isochronous fashion.
This system involves 3G enabled video
telephone and 3G mobile network. 3G
enabled telephone comprises of a video
camera pointing at the user and a video
screen transmitting pictures taken by the
other user’s camera.
The speech and video content are created in
a live fashion from the microphone and
camera input. These are encoded in real-time
by the mobile devices and transmitted in
uplink direction towards the network and the
other end user.
Speech and video data in the opposite
direction (downlink) is conveyed from the
network to source mobile device, which
performs data decoding and display/playback
of video and speech data.

The mobile network carries conversational

multimedia and control traffic in the uplink
and downlink directions, allowing real-time
communication between the two mobile
phone users.
IMPLEMENTATION
To implement mobile video telephony both CS
and PS bearer channel can be used.

The specification for CS mobile video telephony

are based on the ITU-T H.324 standard for
video telephony terminals over circuit switched
channels.

The specification for PS mobile video telephony

are based on the IETF SIP standard for video
telephony over packet-switched channels.
3G-324M is an umbrella protocol that support
real time multimedia services over a wireless
CS network. The architecture comprises of
several sub-protocols that handle error
correction, multiplexing and de-multiplexing,
encoding and decoding of speech and video.

The speech codec used is Adaptive MultiRate

(AMR) that operate at different rate between
4.75kbps and 12.2kbps to provide best
speech quality, while the video codec are
H.263 and MPEG-4.
 Session Initiation Protocol (SIP) in PS
This is an application layer control protocol for creating, modifying,
and terminating sessions with one or more participants. It
performs the logical bound between the media streams of two
video telephony terminals. The protocol stack for PS includes:

Audio & video codec: for speech and video encoding and decoding
(AMR,H.263 & MPEG-4).
RTP: for real-time media data delivery.
RTCP: for QoS monitoring.
SDP: for session properties description.
UDP/TCP: SIP run on them to perform logical bound between the
media streams.
IPv4/v6: RTP runs on it for packet sequence number & time
stamping.
Benefits of Mobile Video Telephony

1.) It affords the deaf people the opportunity

of using sign language to communicate in
video call.
2.) It aids the delivery of medical services over
telecommunication network (tele-medical).
3.) It helps in distance learning (tele-
educational services).
4.) In business live video telephony can be
used as a powerful tool to increase efficiency.
e.g. monitoring of work in progress on a site
of ongoing construction.