Download SIP - OpenLoop.com

Team Members:
Manjiri Ayyar
Pallavi Murudkar
Sriusha Kottalanka
Vamsi Ambati
Girish Satya
LeeAnn Tam
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Introduction to SIP
Overview of functionality
SIP components
SIP protocol layers
SIP call flows
SIP concerns
Demo
Conclusion
Session Initiation Protocol (SIP)
 application layer signaling protocol
 used to create, manage and terminate
sessions in an IP based network.
RFC : 3261
• Circuit is fully established between the two devices before data is sent.
• Less efficient since much of the bandwidth is wasted.
• No fixed path is established between devices
• Data broken into packets.
• Packets may take multiple paths to reach the destination device.
• More efficient.
• VoIP
• Video Conferencing
• Instant Messaging
A typical real-time multimedia session requires
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Session management : Users may move from terminal to
terminal with different capabilities. To set up communication
session between two or more users, a signaling protocol is
needed.
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Media transport : RTP is used for transmitting real-time data
like audio and video.
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End-to-End delivery : Underlying IP layer which connects the
whole world.
SIP is limited to only the setup, modification and termination
of sessions.
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Establishment of user location
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Feature negotiation
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Call management
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Changing features while a session is in progress
All of the other key functions are done with other protocols
The key components in a SIP network are
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SIP Clients : SIP Phones (User-Agents)
SIP servers
SIP PSTN gateways
Application servers (such as media servers)
Application
Transport
Network
Physical/Data
Link
RTSP
SDP
codecs
SIP
RTP
TCP
UDP
IP
Ethernet
DNS(SRV)
Transaction User
Transaction
Transport
Syntax and Encoding
start-line
*message-header
CRLF
[ message-body ]
start-line = Request-Line / Status-Line
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INVITE Requests a session
ACK Final response to the INVITE
OPTIONS Ask for server capabilities
CANCEL Cancels a pending request
BYE Terminates a session
REGISTER Sends user’s address to server
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1XX Provisional 100 Trying
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2XX Successful 200 OK
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3XX Redirection 302 Moved Temporarily
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4XX Client Error 404 Not Found
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5XX Server Error 504 Server Time-out
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6XX Global Failure 603 Decline
Session Registration
Establishment , Termination
RFC 3665
User A
Registrar Server
Location Server
Register sip:[email protected]
[email protected]
Contact 10.18.2.4
200 - OK
Registration binds a
particular device
Contact URI with a
SIP user Address of
Record.
Host2.com
proxy
Host1.com
proxy
Alice
Bob
Invite F1
Invite F2
Invite F4
100 Trying F3
100 Trying F5
180 Ringing F6
180 Trying F7
180 Trying F8
200 OK F9
200 OK F10
200 OK F11
ACK F12
Media Session
Bye F13
200 OK F14
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP
pc33.atlanta.com;branch=z9hG4bK776asdhds
Max-Forwards: 70
To: Bob <sip:[email protected]>
From: Alice <sip:[email protected]>;tag=1928301774
Call-ID: [email protected]
CSeq: 314159 INVITE
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 142
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SIP/2.0 200 OK
Via: SIP/2.0/UDP server10.biloxi.com
;branch=z9hG4bKnashds8;received=192.0.2.3
Via: SIP/2.0/UDP bigbox3.site3.atlanta.com
;branch=z9hG4bK77ef4c2312983.1;received=192.0.2.2
Via: SIP/2.0/UDP pc33.atlanta.com ;branch=z9hG4bK776asdhds
;received=192.0.2.1
To: Bob <sip:[email protected]>;tag=a6c85cf
From: Alice <sip:[email protected]>;tag=1928301774
Call-ID: [email protected]
CSeq: 314159 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 131
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Security
◦ Authentication of signaling data using HTTP digest
authentication
◦ TLS usage (over TCP)
◦ Usage of IPSec (SIP VPN Scenario)
◦ Use SecureRTP for Media
◦ Use S/MIME to enable mechanisms like public key
distribution, authentication, integrity and confidentiality
of SIP signaling data
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Quality of Service
◦ Latency, network delays (upper bound is 150ms)
◦ Jitter ( refers to non-uniform delays )
◦ Packet Loss
◦ Power Failure and Backup Systems
◦ Interoperability
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User Agents used : Yahoo Messenger
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Call Scenarios Covered:
◦ Register
◦ Call Establishment
◦ Call Termination