Session Initiation Protocol Team Members: Manjiri Ayyar Pallavi Murudkar Sriusha Kottalanka Vamsi...
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Transcript of Session Initiation Protocol Team Members: Manjiri Ayyar Pallavi Murudkar Sriusha Kottalanka Vamsi...
Session Initiation Protocol
Team Members:
Manjiri AyyarPallavi Murudkar
Sriusha KottalankaVamsi AmbatiGirish SatyaLeeAnn Tam
Agenda Introduction to SIP Overview of functionality SIP components SIP protocol layers SIP call flows SIP concerns Demo Conclusion
Introduction
Session Initiation Protocol (SIP) application layer signaling protocol used to create, manage and terminate
sessions in an IP based network. RFC : 3261
Circuit switched Network
• Circuit is fully established between the two devices before data is sent.
• Less efficient since much of the bandwidth is wasted.
Packet switched network
• No fixed path is established between devices• Data broken into packets.• Packets may take multiple paths to reach the destination device.• More efficient.
SIP applications
• VoIP
• Video Conferencing
• Instant Messaging
Multimedia session in a packet switched networkA typical real-time multimedia session requires
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.
Media transport : RTP is used for transmitting real-time data like audio and video.
End-to-End delivery : Underlying IP layer which connects the whole world.
SIP functionalitySIP is limited to only the setup, modification and termination of sessions.
Establishment of user location Feature negotiation
Call management
Changing features while a session is in progress
All of the other key functions are done with other protocols
SIP components
The key components in a SIP network are
SIP Clients : SIP Phones (User-Agents) SIP servers SIP PSTN gateways Application servers (such as media
servers)
SIP Network
Application
Transport
Network
Physical/Data Link
Ethernet
IP
TCP UDP
RTSP SIP
SDP codecs
RTP DNS(SRV)
Transport
Transaction
Transaction User
Syntax and Encoding
start-line*message-headerCRLF[ message-body ]start-line = Request-Line / Status-Line
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
1XX Provisional 100 Trying
2XX Successful 200 OK
3XX Redirection 302 Moved Temporarily
4XX Client Error 404 Not Found
5XX Server Error 504 Server Time-out
6XX Global Failure 603 Decline
Session Registration Establishment , TerminationRFC 3665
User A Registrar Server Location Server
Register sip:[email protected]
[email protected] 10.18.2.4
200 - OK Registration binds a particular device Contact URI with a SIP user Address of Record.
Alice Host1.com proxy
Host2.com proxy Bob
Invite F1Invite F1
Invite F2Invite F2Invite F4Invite F4
100 Trying F5100 Trying F5100 Trying F3100 Trying F3
180 Ringing F6180 Ringing F6
180 Trying F8180 Trying F8
180 Trying F7180 Trying F7
ACK F12ACK F12
200 OK F9200 OK F9
200 OK F11200 OK F11200 OK F10200 OK F10
Media Session
Bye F13Bye F13
200 OK F14200 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]: application/sdp Content-Length: 142
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
SIP Concerns 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
SIP Concerns…contd Quality of Service
◦ Latency, network delays (upper bound is 150ms)
◦ Jitter ( refers to non-uniform delays )
◦ Packet Loss
◦ Power Failure and Backup Systems
◦ Interoperability
Demo
User Agents used : Yahoo Messenger
Call Scenarios Covered:
◦Register
◦Call Establishment
◦Call Termination