SIP Fundimentals

Vini Meiriska
1515031101

Dennis Baron, January 15, 2008

np160 Page 1
 What’s SIP

• IETF Standard defined by RFC 3261

• “The Session Initiation Protocol (SIP) is an application-layer
control (signaling) protocol for creating, modifying and
terminating sessions with one or more participants.”
• Can be used for voice, video, instant messaging, gaming,
etc., etc., etc.
• Uses URIs for addressing – single communications identity
– mailto:dbaron@MIT.edu for email

– xmpp:dbaron@MIT.EDU for instant messaging

– sip:dbaron@MIT.EDU for voice and video

• Username replaced by numbers for telephone applications

Dennis Baron, January 15, 2008

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 Where’s SIP

SDP codecs

Application RTSP SIP RTP DNS(SRV)

Transport TCP UDP

Network IP

Physical/Data Link Ethernet

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 SIP Components

• User Agents
– Clients – Make requests
– Servers – Accept requests

• Server types
– Redirect Server
– Proxy Server
– Registrar Server
– Location Server

• Gateways

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 SIP Trapezoid

DNS Location
Server Server

DNS
Registrar
SIP

Outbound Inbound
Proxy Proxy
SIP SIP SIP

SIP
Originating Terminating
User Agent RTP User Agent

Dennis Baron, January 15, 2008

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 SIP Triangle ?

DNS Location
Server Server

DNS
Registrar

Inbound
Proxy
SIP SIP SIP

SIP
Originating Terminating
User Agent RTP User Agent

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 SIP Peer to Peer !

SIP
Originating Terminating
User Agent RTP User Agent

Back-to-Back User Agent

B2BUA

SIP SIP

Originating Terminating
User Agent RTP User Agent

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 SIP Methods

• 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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 SIP Responses

• 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

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 SIP Flows - Basic
User User
A B

“Calls”
INVITE: sip:18.10.0.79
18.18.2.4

180 - Ringing Rings

200 - OK Answers

ACK

Talking RTP Talking

Hangs up BYE

200 - OK

Dennis Baron, January 15, 2008

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 Session Description Protocol
• IETF RFC 2327
• “SDP is intended for describing multimedia sessions for the
purposes of session announcement, session invitation, and
other forms of multimedia session initiation.”
• SDP includes:
– The type of media (video, audio, etc.)
– The transport protocol (RTP/UDP/IP, H.320, etc.)
– The format of the media (H.264 video, MPEG video,
etc.)
– Information to receive those media (addresses, ports,
formats and so on)

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 CODECs

• Audio
– G.711
• 8kHz sampling rate
• 64kbps
– G.729
• 8kHz sampling rate
• 8kbps
• Voice Activity Detection

• Video
– H.264
• MPEG-4
– H.263

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 SIP Flows - Registration
User Registrar Location
B MIT.EDU MIT.EDU

REGISTER: sip:dbaron@MIT.EDU

401 - Unauthorized

REGISTER: (add credentials)

sip:dbaron@MIT.EDU
Contact 18.10.0.79
200 - OK

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 SIP Flows – Via Proxy
User Proxy User
A MIT.EDU B

“Calls” dbaron
INVITE: sip:dbaron@MIT.EDU
@MIT.EDU
INVITE:sip:dbaron@18.10.0.79
100 - Trying
180 - Ringing Rings
180 - Ringing

200 - OK Answers
200 - OK

ACK

Talking RTP Talking

Hangs up BYE

200 - OK

Dennis Baron, January 15, 2008

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 SIP Flows – Via Gateway
User Proxy Gateway
A MIT.EDU 38400

“Calls” joeuser
INVITE: sip:joeuser@MIT.EDU
@MIT.EDU
INVITE: sip:38400@18.162.0.25
100 - Trying Rings
180 - Ringing
180 - Ringing
Answers
200 - OK
200 - OK

ACK
ACK

Talking RTP Talking

Hangs up BYE
BYE

200 - OK

200 - OK

Dennis Baron, January 15, 2008

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 SIP Standards

Just a sampling of IETF standards work…

IETF RFCs http://ietf.org/rfc.html

• RFC3261 Core SIP specification – obsoletes RFC2543
• RFC2327 SDP – Session Description Protocol
• RFC1889 RTP - Real-time Transport Protocol
• RFC2326 RTSP - Real-Time Streaming Protocol
• RFC3262 SIP PRACK method – reliability for 1XX messages
• RFC3263 Locating SIP servers – SRV and NAPTR
• RFC3264 Offer/answer model for SDP use with SIP

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 SIP Standards (cont.)
• RFC3265 SIP event notification – SUBSCRIBE and NOTIFY
• RFC3266 IPv6 support in SDP
• RFC3311 SIP UPDATE method – eg. changing media
• RFC3325 Asserted identity in trusted networks
• RFC3361 Locating outbound SIP proxy with DHCP
• RFC3428 SIP extensions for Instant Messaging
• RFC3515 SIP REFER method – eg. call transfer
• RFC4474 Authenticated Identity Management

• SIMPLE IM/Presence - http://ietf.org/ids.by.wg/simple.html

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 IS&T Services
• MITvoip
– Desktop VoIP telephones to replace traditional 5ESS
telephones
– New voice mail system
– Web interface for user control
– Transition over 2 to 2.5 years

• Personal SIP accounts

– Bring your own devices/software
– Limited support

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 Soft and Hard SIP Clients
• “Hard phones”

• “Soft phones”

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