SIP Technology Overview:

Session Initialization
Protocol
TP-0062.1, May 2000, Slide 1
 Objectives
 To understand purpose, components, and
operation of SIP
 Upon completion of this section, you will be able
to:
• Define purpose of SIP
• Identify components used by SIP
• Explain how SIP works
• Differentiate between good and problematic SIP
responses
• Identify the well known port used by SIP
• Understand major differences between SIP and
H.323

TP-0062.1, May 2000, Slide 2

 What is SIP?
 Session Initialization Protocol (SIP) is an
application layer control (signalling) protocol for
creating, modifying and terminating sessions
with one or more participants
 These sessions include Internet multimedia
conferences, distance learning, Internet
telephone calls, and multimedia distribution
 SIP can invite both persons and "robots," (such
as a media storage service) to participate in a
call
 SIP is defined in RFC 2543

TP-0062.1, May 2000, Slide 3

 What Can SIP Be Used For?
Multipoint Sessions
Voice
Server

SIP IP

Invite to Voice Session

 To initiate sessions
 To invite members to sessions advertised by
other means
 To initiate multi-party calls using a multipoint
control unit (MCU)
TP-0062.1, May 2000, Slide 4
 SIP Name Mapping

Redirect

SIP
Call to SAM Server
IP
Redirect to Mobiles

 SIP transparently supports name mapping and

redirection services, allowing implementation of
ISDN & intelligent network telephony subscriber
services (which includes services such as personal
mobility)
TP-0062.1, May 2000, Slide 5
 What Else is SIP Used For?
 SIP is a flexible protocol designed to be part of
the overall IETF multimedia data and control
architecture
 SIP may be used in conjunction with other call
setup & signaling protocols
 SIP can be used to determine caller reachability
(via either H.323 or PSTN)
 SIP can be used between 2 Internet telephone
gateways to set up calls between the 2 PSTNs

TP-0062.1, May 2000, Slide 6

 What Doesn’t SIP Do?
 Offer conference control services
 Prescribe how a conference is to be managed
 Allocate multicast addresses
 Reserve resources, but it does convey the
information necessary to do this to the invited
system

TP-0062.1, May 2000, Slide 7

 What Does SIP Support?
 5 facets of establishing and terminating
multimedia communications:
1. User location: determination of end system to be
used for communication
2. User capabilities: determination of media and media
parameters to be used
3. User availability: determination of willingness of
called party to engage in communications
4. Call setup: ringing, establishment of call parameters
at both called and calling party
5. Call handling: including transfer and termination of
calls

TP-0062.1, May 2000, Slide 8

 SIP Operation
 Callers & recipients are identified by SIP
addresses
 A caller first locates the appropriate server, then
sends a SIP request (probably an INVITE)
 In the perfect world, the request arrives at its
destination when the client accepts the call by
returning a SIP response code 200
 Originating caller then sends an ACK to recipient

TP-0062.1, May 2000, Slide 9

 Basic Operation
End Proxy Location User Agent End
User INVITE Server Server Server User
mj@bulls.org
mj

mj@bbThor

INVITE
mj@bbThor

RING

200 OK

200 OK
ACK ACK

TP-0062.1, May 2000, Slide 10

 How Does SIP Connect Session Members?
 SIP can communicate via:
• Multicast
• A mesh of unicast relationships
• A combination of multicast and unicast

TP-0062.1, May 2000, Slide 11

 SIP Addressing
 Objects addressed by SIP are users at hosts
 Users are identified by a SIP Uniform Resource
Locator (URL) [user@host]
• User part is a user name or a telephone number
• Host part is a domain name or IP address
 SIP addresses may be obtained out-of-band,
learned via media gateways, recorded during
earlier conversations, or guessed (since they’re
often similar to E-mail addresses)

TP-0062.1, May 2000, Slide 12

 What is the Appropriate Server?
 There are several different SIP servers, each with
their own purpose
 There are:
• User agent servers
• Proxy servers
• Redirect servers
• Registrars
 There is also something called a location server
running a location service which may be co-
located with a SIP server

TP-0062.1, May 2000, Slide 13

 User Agent Servers (UAS)
 Application which contacts user when a SIP
request is received
 UAS then returns a response on behalf of user
 Response either accepts, rejects, or redirects the
request

SIP Call Request User “Call for you?”

Agent
Call Accepted Server “I’ll accept.”

TP-0062.1, May 2000, Slide 14

 Proxy Server
 Intermediary program that acts as both a server and a
client to make requests on behalf of other clients
 A proxy interprets a request, and if necessary, rewrites
request message before forwarding it

User
Agent 1

Originates SIP
Requests
Proxy User
Server Agent X
Responds for 1 and X
Answers on behalf
of user.
TP-0062.1, May 2000, Slide 15
 Redirect Server
 Redirects calls to new User Agent
location
 Provides router Redirect
functionality Server
 A server that accepts a SIP
request, maps address
into 0 or more new
addresses and returns User Agent
those addresses to client
Proxy
 Unlike a proxy server, a Agent
redirect server does not
initiate its own SIP request
PSTN
 Unlike a user agent server,
a redirect server does not
accept calls
TP-0062.1, May 2000, Slide 16
 Registrar
 Allows users to register
their presence
 A server that accepts Proxy Server
REGISTER requests
 Typically co-located
with a proxy or
redirect server Register
Server
 May offer location
services Redirect
Server

TP-0062.1, May 2000, Slide 17

 Locating a SIP Server
 A client sends a request either to a Universal
Resource Identifier (Request-URI), or a locally
configured SIP proxy server
 If sending to a Request-URI, client must
determine IP address, protocol, and port number
of destination server
• DNS is used to obtain an IP address
• Client should try to contact server at a specified port
or well known port 5060
• If no protocol is specified, UDP is assumed
• If UDP fails, TCP is attempted

TP-0062.1, May 2000, Slide 18

 Server Name
 There are no rules for naming SIP servers
 RFC 2543 encourages the naming convention of
sip.domainname (i.e., sip.radcom-inc.com)
 Clients may (and probably should) cache
successful DNS queries

Hey, what’s
your name?

TP-0062.1, May 2000, Slide 19

 SIP Transactions
 Consists of a SIP request with the appropriate
response
 To identify which SIP request a response is for,
the following fields should be identical:
• Call-ID - uniquely identifies a particular invitation
• Cseq (command sequence number) - a unique
number that is incremented for each command with
the same call-ID
• To
• From
• Tag (only present when more than one Via header
field is present)

TP-0062.1, May 2000, Slide 20

 INVITE
 Asks a recipient to join a conference call or
establish a 2-party call
 Includes a session description (probably using
SDP); the session description enumerates media
types & formats
 If recipient agrees, caller sends an ACK & returns
a description listing media it wishes to use
• For multicast sessions, recipient should only return
a session description if it is unable to receive media
indicated

o u a re
Y
r d ia l ly
co .
it e d . .
TP-0062.1, May 2000, Slide 21
nv
 SIP Status Codes

Code Reason-Phrase Definition

1xx Informational Request received, continuing to process the request
2xx Success Action was successfully received, understood, and accepted
3xx Redirection Further action needed to complete request
4xx Client Error Request contains bad syntax or cannot be fulfilled at this server
5xx Server Error Server failed to fulfill an apparently valid request
6xx Global Failure Request cannot be fulfilled at any server

TP-0062.1, May 2000, Slide 22

 INVITE With a Proxy Server
End Proxy Location User Agent End
User Server Server Server User
INVITE
mj@bulls.org
mj

mj@bbThor

INVITE
mj@bbThor

RING

200 OK

200 OK
ACK ACK

bulls.org
TP-0062.1, May 2000, Slide 23
 INVITE With a Redirect Server
End Redirect Location User Agent End
User Server Server Server User
INVITE
mj@bulls.org
mj

302 MOVED mj@bbThor

mj@bbThor

ACK

INVITE
mj@bbThor.bulls.org
RING

200 OK

ACK
bulls.org
TP-0062.1, May 2000, Slide 24
 Summary

Voice/video
educational
content

SIP
Servers

SIP initialization
and session
established

TP-0062.1, May 2000, Slide 25

 SIP vs. H.323: Major Differences
 Call setup: SIP is like H.323 V3
 Supplementary services: H.450. 7/8 – only in
H.323
 Third-party control (ability for a party to set up a
call between 2 other parties without participating
in the call): only in SIP
 Capability exchange (what a terminal can receive
and decode): limited in SIP, compared to H.245
 Call setup delay (number of round trips needed
for establishing a call): H.323 V2 has Fast Start,
and H.323 V3 is almost the same as SIP

TP-0062.1, May 2000, Slide 26

 SIP vs. H.323: Major Differences (cont.)
 Complexity: H.323 is rather complex protocol
(H.225, H.245, H.450.X…)
 Flexibility: H.323 uses ASN.1 extension
mechanism (non-standard field), while SIP
provides easy-to-use extension space
 Interoperability among versions: H.323 is always
backward compatible (V1 -> V2 -> V3); SIP does
not guarantee backward compatibility

TP-0062.1, May 2000, Slide 27

 SIP vs. H.323: Major Differences (cont.)
 Interoperability with other signalling protocols:
H.225 is Q.931-based protocol, which is used in
ISDN and ISUP/SS7; no SIP support
 Ease of implementation: H.323 uses ASN.1
(binary) encoded signaling; SIP messages are
text-based

TP-0062.1, May 2000, Slide 28

 RADCOM’s SIP Support
 RADCOM has a wide range of SIP test solutions
that includes SIP simulators and analyzers
• For more information visit our website :
www.radcom.com

TP-0062.1, May 2000, Slide 29