Side_1_360

(Dana P.) #1

a page is fetched, the local cache can be exam-
ined to see if the page is there and if the page is
still up to date. This allows a more rapid view of
the corresponding pages as the information does
not have to be transferred.


Considering that human end users are commonly
involved and that significant amounts of data are
transferred, ways to improve the service perfor-
mance are strived for. One approach is to store
some of the objects on the page on a different
server more local to the user. This server may be
dynamically selected based on current load and
performance pattern. This may be called traffic
directingas requests are directed to other
servers.


Another aspect is to balance the loads among
the servers, e.g. to speed up the delivery of Web-
pages. How to balance the load in a distributed
environment is a complex matter; one way being
to introduce a load balancer which can be imple-
mented in different terminals/servers.


A proxy sever can be seen as a gateway having
multiple functions. One function can be that it
accepts HTTP requests and translates these into
other protocols, e.g. FTP. Another function is
that the proxy server may implement a cache. A
third function is access control, both for requests
going out to the rest of the networks and for
responses that arrive, i.e. a firewall function.
Hence, a proxy is commonly present for a com-
pany network supporting Web browsing.


7.4 Supporting Telephony

7.4.1 Protocols and Servers
The Session Initiation Protocol (SIP) is a proto-
col elaborated by IETF activities used for estab-
lishing, modifying and releasing real-time calls
and conferences over IP-based networks. Each
session may include different traffic flows, such
as audio and video. SIP is a text-based and gen-
eral-purpose protocol. An alternative is to use
the H.323 architecture as described by ITU-T.


Broadly speaking, SIP may be thought of as the
call control protocol of an IP session. The basic
SIP architecture may include a location data
base that allows users to be contacted at the
locations where they are registered. For this
five aspects are considered, ref. [RFC2543]:


  • User location: determination of the end system
    to be used for communication;

  • User capabilities: determination of the media
    and media parameters to be used;

  • User availability: determination of the willing-
    ness of the called party to engage in communi-
    cations;

  • Call set-up: “ringing”, establishment of call
    parameters at both called and calling party;

  • Call handling: including transfer and release
    of calls.


SIP is part of the multimedia data and control
architecture as depicted in Figure 31. RSVP is
used for reserving network resources, the real-
time transport protocol (RTP) is used for trans-
porting real-time data and providing feedback,
the real-time streaming protocol (RTSP) is used
for controlling delivery of streaming media, the
session announcement protocol (SAP) for adver-
tising multimedia sessions via multicast, and the
session description protocol (SDP) for describ-
ing multimedia sessions. However, it is stated
that SIP does not depend on either of these.

The Real-Time Protocol (RTP) has been men-
tioned, which has emerged as a commonly used
protocol for real-time traffic flows in IP-based
networks. RTP is a protocol that provides identi-
fication of media type and synchronisation infor-
mation (time stamps). These allow individual
packets to be reconstructed by a receiver. Addi-
tional information is needed to support flow con-
trol and management of the traffic flows. Here
the RTP Control Protocol (RTCP) has been

Figure 24 Example of SIP
message

SDP description

v= 0
o = tom.jones 3546342323 6434236545 IN IP4 telenor.com
s = Session SDP
e = [email protected]
e = IN IP4 193.291.192
t = 0 0
m = audio 9150 RTP/AVP 0
a = rtpmap:0 PCMU/8000

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0.UDP mach.tel:5060
From: Tom Jones <sip:[email protected]>
To: Cliff Rich <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 INVITE
Subject: Call
Contact: Tom Jones <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 160

SIP header UDP header IP header
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