on a packet switched network, as described ear-
lier. This means that there are no guarantees that
the packets will arrive according to the same dis-
tribution as they are put on the network, that the
packets are received in the right order or that all
packets are delivered at all. Applications typi-
cally run RTP on top of UDP to make use of the
services provided by UDP. The sequence number
in RTP is used to reconstruct the sender’s packet
sequence or to determine a proper location of a
packet in a coded packet stream.
The Real-time Control Protocolis based on the
foundation of its packets being one among oth-
ers. RTCP packets are regularly inserted into the
packet stream and transmitted as ordinary pack-
ets. These probe packets are then measured and
an estimate of the behaviour of the transmitted
service is obtained. As the services introduced
gradually become more time sensitive a certain
QoS has to be maintained.
The Resource Reservation Protocolis designed
to address those requirements. When an applica-
tion needs a certain QoS grade for its service it
consults the RSVP to request support for that
level of QoS. The control packets could be sent
directly inside IP packets, which are encapsu-
lated by the UDP. One of the drawbacks of this
protocol is that to be able to support, provide and
promise the requested QoS grade the protocol
has to be employed in all routers in the network,
or at least all routers along the path of the con-
nection. The RSVP is however not a routing pro-
tocol, it is only concerned with the QoS of those
packets forwarded by the network’s routing pro-
tocol. The protocol requests that the receiver and
the links along the connection path are reserved
to support the data flow.
Finally, the Real-time Streaming Protocol,
which is an application layer protocol, controls
the delivery of the real time packet stream.
Examples of services supported by RTSP are
to accept additions of media to already existing
presentations, as additional media becomes
available, and retrievals of media from media
servers. The RTSP protocol is in structure rather
similar to the HTTP (Hypertext Transfer Proto-
col), which means that extensions made to
HTTP also, in most cases, could be deployed to
RTSP. Among the differences between the pro-
tocols the out of band data transfer implemented
by RTSP should be enlighten.
Conclusions
After the brief introduction of different network
switching principles and the concept of sending
voice in packets the paper is devoted to VoIP
issues. The structure of the VoIP packet as well
as some implementation issues were described.
The protocols supporting VoIP transmissions,
ensuring QoS and other real time and streaming
problems were generically described. The
requirements for voice coders were also com-
mented on. Even though some problems still
exist for VoIP to be an acceptable service with
QoS for ordinary telephony, it is slowly being
implemented in the networks. It is a very cheap
solution for international voice communication,
if the lower quality is acceptable. This also indi-
cates that it is, so far, most applicable to direct
computer communications. The VoIP protocols
are by themselves adequate for a good connec-
tion, it is the lower layer protocols, i.e. IP, that
need to have some further QoS assurance param-
eters implemented. The described H.323 family
of protocols is designed to provide for interoper-
ability, and several working groups within dif-
ferent standardization organisations are working
towards that end.References
1 ITU. One-way transmission time.Geneva,- (ITU-T G.114.)
2 Goodman, B. Internet Telephony and
Modem Delay. IEEE Network Magazine, 13
(3), 8–16, 1998.3 Reynolds, J, Postel, J. Assigned Numbers.- (RFC 1700.)
4 ITU. Methods for Subjective Determination
of Transmission Quality.Geneva, 1996.
(ITU-T P.800.)5 ITU. Dual rate speech coder for multimedia
communications transmitting at 5.3 and 6.3
kbps.Geneva, 2000. (ITU-T G.723.1.)6 Handley, M et al. SIP: Session Initiated Pro-
tocol.1999. (RFC 2543.)7 Schulzrinne, H, Rosenberg, J. The Session
Initiation Protocol: Providing Advanced
Telephony Services Across the Internet. Bell
Labs Technical Journal, 3 (4), 144–160, 1998.8 IETF Media Gateway Control Working
Group (MEGACO). (2001, August 12)
[online] – URL: http://www.ietf. org or
URL: http://standards.nortelnetworks.
com/archives/megaco.html9 Schulzrinne, H et al. Real Time Streaming
Protocol (RTSP). 1998. (RFC 2326.)10 Deering, S. Host Extensions for IP Multicas-
ting.1989. (RFC 1112.)11 Deering, S, Hinden, R. Internet Protocol
Version 6 (Ipv6) Specification.1998. (RFC
2460.)