Session Initiation Protocol

(SIP)

by
B Pavan Kumar
SIP based VoIP Architecture
I
N
Application
T
3pcc Services
E eMail CPL
CPL
LDAP Oracle XML
L
L
I
G SIP Proxy, Registrar
E & Redirect Servers
N
T SIP

SIP SIP
S SIP User PSTN
E Agents (UA)
R
CAS or PRI
V
I RTP
C (Media)
E 2
S Legacy PBX
Basic SIP Call-Flow
SIP UA1 SIP UA2

INVITE w/ SDP for Media Negotiation

100 Trying

180/183 Ringing w/ SDP for Media Negotiation

MEDIA

200 OK

ACK

MEDIA

BYE

200 OK
3
SIP Call Flow with Proxy Server
User Agent 1 Proxy Server User Agent 1

Register Register
OK (200) OK (200)
Invite Invite
Trying (100)
Ringing (180)
Ringing (180)
OK (200)
OK (200)
ACK ACK

RTP/RTCP media
channels
4
VoIP Migration
 Step1: IPPBX deployments in Enterprises

PSTN
Network

Customer Premises Customer Premises

IP Core
Network

- Large enterprises will handle VOIP calls directly DNS Server for URL resolution
- PSTN connectivity provided by Media Gateways
- Regulation can not stop spammers outside USA
(similar to SMTP spam)

6
 STEP 2: Hosted IP Centrex
FW, NAT, VoIP service provided by Carrier Networks

Softswitches, MGW
VoIP Proxy Server, SGW Internet
SGC, VoIP Centrex Server,

Carrier Network

Customer Premises
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 Step 3: Carrier VoIP Network

VoIP Trunk

Softswitches, MGW Internet

VoIP Proxy Server, SGW
SGC, VoIP Centrix Server,

Carrier Network

- VoIP FW, NAT and Security provided by Carriers

Customer Premises

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SIP Architecture
The Popularity of SIP

Originally Developed in the MMUSIC
 A separate SIP working group
 RFC 2543, RFC 3261
 Many developers

SIP + MGCP/MEGACO
 The VoIP signaling in the future

“back-off” or SIPit (SIP Interoperability
Tests)
 Test products against each other
 Will be hosted by ETSI
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SIP Architecture

A signaling protocol
 The setup, modification, and tear-down of
multimedia sessions

SIP + SDP
 Describe the session characteristics

Separate signaling and media streams

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SIP Network Entities

Clients
 User agent clients
 Application programs sending SIP requests

Servers
 Responds to clients’ requests

Clients and servers may be in the same
platform

Proxy
 Acts as both clients and servers

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Four types of servers
 Proxy servers

Handle requests or forward requests to other
servers

Can be used for call forwarding

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 Redirect servers

Map the destination address to zero or more new
addresses

Do not initiate any SIP requests

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 A user agent server

Accept SIP requests and contacts the user

The user responds → an SIP response

A SIP device

E.g., an SIP-enabled telephone
 A registrar

Accepts SIP REGISTER requests
 Indicating the user is at a particular address

Typically combined with a proxy or redirect
server

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 SIP Call Establishment

It is simple
 A number of interim responses

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SIP Advantages
 Attempt to keep the signaling as simple
as possible
 Offer a great deal of flexibility
 Various pieces of information can be
included within the messages

Including non-standard information

Enable the users to make intelligent
decisions
 The user has control of call handling

No need to subscribe call features
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 Call Completion to Busy Subscriber service

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Via contains the address (e.g., pc33.atlanta.com)

Contact contains a SIP or SIPS URI that represents a
direct route to contact the called party, usually
composed of username at a fuly qualified domain name
(FQDN). While the FQDN is preferred, many end systems
do not have registered domain names, so IP addresses
are permitted. While Via header field tells other
elements where to send response, the Contact header
field tells other elements where the called party can be
reached directly.

In a response, Via, To, From, Call-ID, and CSeq header
fields are copied from the INVITE request.

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In addition to DNS and location service
lookups, proxy servers can make flexible
“routing decisions” to decide where to send a
request. For example, if Bob’s SIP phone
returned 486 (busy) response, the biloxi.com
proxy server could proxy the INVITE to Bob’s
voicemail server. A proxy server can also send
an INVITE to a number of locations at the
same time. This type of parallel search is
known as forking.

After learning the end point addresses, the
end points can communicate directly
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Overview of SIP Messaging
Syntax

Text-based
 Similar to HTTP

SIP messages
 message = start-line
*message-header CRLF
[message-body]
 start-line = request-line | status-line

Request-line specifies the type of request

The response line
 The success or failure of a given request

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Message headers
 Additional information of the request or response
 E.g.,

The originator and recipient

Retry-after header

Subject header

Message body
 Describe the type of session

The media format
 SDP, Session Description Protocol
 Could include an ISDN User Part message
 Examined only at the two ends
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SIP Requests

method SP request-URI SP SIP-version CRLF

request-URI
 The address of the destination

Methods
 INVITE, ACK, OPTIONS, BYE, CANCLE, REGISTER

extensions: INFO, REFER, UPDATE, …
 INVITE

Initiate a session

Information of the calling and called parties

The type of media

IAM (initial address message) of ISUP

ACK only the final response
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 BYE

Terminate a session

Can be issued by either the calling or called
party
 Options

Query a server as to its capabilities
 A particular type of media
 The response if sent an INVITE
 CANCEL

Terminate a pending request

E.g., an INVITE did not receive a final response

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 REGISTER

Log in and register the address with a SIP server

“all SIP servers” – multicast address
(224.0.1.1750)

Can register with multiple servers

Can have several registrations with one server
 INFO

RFC 2976

Transfer information during an ongoing session
 DTMF digits
 account balance information
 midcall signaling information generated in
another network
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SIP Responses

SIP version SP status code SP reason-phrase CRLF

reason-phrase
 A textual description of the outcome
 Could be presented to the user

status code
 A three-digit number
 1XX Informational
 2XX Success (only code 200 is defined)
 3XX Redirection
 4XX Request Failure
 5XX Server Failure
 6XX Global Failure
 All responses, except for 1XX, are considered final 26

Should be ACKed
“One number” service

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SIP Addressing

SIP URLs (Uniform Resource Locators)
 user@host
 E.g.,

sip:collins@home.net

sip:3344556789@telco.net
 Supplement the URL

sip:3344556789@telco.net;user=phone
 sip:user:password@host:port;uri-parameters?headers

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Message Headers

Provide further information about the message
 information elements

E.g.,
 To:header in an INVITE

The called party
 From:header

The caling party

Four main categories
 General, request, response, and entity headers
 A list in Table 5-2
 Mapping in Table 5-3

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General Headers
 Used in both requests and responses
 Basic information

E.g., To:, From:, Call-ID:, …
 Contact:

A URL for future communication

May be different from the From: header
 Requests passed through proxies

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Request Headers
 Apply only to SIP requests
 Addition information about the request or the client
 E.g.,

Subject:

Priority:, urgency of the request

Authorization:, authentication of the request
originator

Response Headers
 Further information about the response
 E.g.,

Unsupported:, features

Retry-After
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Entity Header
 Session information presented to the user
 Session description, SDP

The RTP payload type, an address and port
 Content-Length, the length of the message body
 Content-Type, the media type of the message
 Content-Encoding, for message compression
 Content Disposition,
 Content-Language,
 Allow, used in a Request to indicate the set of
methods supported
 Expires, the date and time

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 Example of SIP Message
Sequences

Registration
 Via:
 Call-ID:

host-specific
 Content-Length:

Zero, no msg body
 Cseg:

Avoid ambiguity
 Expires:

TTL

0, unreg
 Contact:

* 33
Invitation

A two-party call
 Subject:

optional
 Content-Type:

application/sdp

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35
Termination of a Call

Cseq:
 Has changed

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