SIP

Anna Sfairopoulou Page 1

 What we will see...
• Signalling vs Media
• SIP standarization and design principles
• Message syntax
• SIP addressing
• SIP methods
– Request messages
– Response messages
• SIP entities (servers)
• Examples

Anna Sfairopoulou Page 2

 Why SIP
• SIP  VoIP + WLAN  competition to
cellular telephony?
• IMS component
• P2PSIP
• Application layer mobility using SIP

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 The simplest voice call
Self-written Java code:
Self-written C code:
•Open socket
•Open socket
•Wait for UDP packets
•Take voice samples from microphone
•Playback voice samples
•Send voice samples in UDP packets
Bob
Alice

Internet

[Link]
Knows address [Link] [Link]
Knows port 5060 Listening on port 5060
Expecting UDP
Knows UDP
Expecting CODEC PCM A-law
Knows CODEC PCM A-law
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 What if...
• Alice doesn't know Bob's IP address?
• Alice uses a different codec?
• Bob is busy in a previous call?
• Bob is absent?
• Other media desired (e.g. Video)?
• Cesar would like to join in?

Some form of call control (a.k.a. SIGNALING) is

needed!

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Think about the road system...

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 Media vs Signaling
• To interchange user data, transport
protocols are used
– UDP
(+ RTP on top)
– TCP

• To control that interchange, signaling

protocols are used
– RSVP
– SIP
– RTSP

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 Signaling serves to...
• Set up and tear down VoIP calls/sessions
• Reserve bandwidth among routers
• Establish secure connections over the
Internet
• Interchange routing information
• Inform of network malfunctions
• Configure network equipment remotely
• Etc.

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The ATM cube revisited

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 Layers vs Planes
• Layering: Structures how data transport is
done in a network
• Planes: Define what purpose a data transfer
serves:
– User plane: Transfer of user info
– Control plane: Control the transfer of user info
– Management plane: Check and update status of
network elements (routers, servers, etc.)

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 The purpose of signaling
• Interchange of control information for the
management of traffic aggregates and of the
network as a whole
• Signaling protocols represent a parallel
network for the purpose of network and
traffic control
The complexity and the intelligence of the
network lies mainly here!

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The real “simplest” voice call

Anna Sfairopoulou Page 12

The real “simplest” voice call

Database (MySQL)

DNS (Named)

SIP Express
Server (SER)

Twinkle

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 Signalling
• Signalling serves to control user data
transfer across the network
– Data transfer is easy, signaling is hard!

• SIP is a session-layer signaling protocol

• RTP is a session-layer data transfer protocol
• UDP is the transport protocol mostly used to
transport both

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 SIP standardization
• First designed by Henning Schulzrinne and
Mark Handley starting in 1996
• Standardized in IETF RFC 3261
– See also RFC 3551, 2327, 3219, 3761, 3525,
3398 ...
• Decentralized design
– The intelligence lies on the nodes
• Ongoing process
– Extensions
– Services

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 SIP design principles (I)
• Modelled after HTTP
– Text-based
• Based on transactions
– Request / response
• Simple
– Less than 10 methods / packet types
• Only for call control
– No QoS, no media transfer, no application-specific
functions

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 SIP design principles (II)
• Session-oriented
– Can manage any session-based service
• Online gaming
• Videoconferencing
• Virtual reality
• Chat
• Etc

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 The SIP-related protocol “Zoo”
• Loosely coupled to other Internet protocols
– SDP, DNS, SMTP, HTTP, RSVP, ...
• Defines complementary protocols to enable
special services
– TRIP, ENUM, SPIRITS, PINT, ...
• Any VoIP network needs a combination of
protocols to work

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 VoIP protocol stack
Presentation Signaling Media Transport Support
Layer
MPEG, H.261, etc.

Session SIP RTP RTCP, TRIP, etc.

Transport TCP / UDP / SCTP? UDP TCP / UDP?

Network IP

Datalink Fast/GigEth, ATM, 802.11 MAC/LLC, etc.

Physical DSL, Phy GigEth,802.11 PHY etc.

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 SIP fundamentals
• Which messages do we need to control
a VoIP call?
– Let's try to find out...

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 A basic SIP call
Alice Bob
| | INVITE sip:bob@[Link] SIP/2.0
| INVITE | From: Alice <sip:alice@[Link]>
|----------------------->| To: Bob <sip:bob@[Link]>
| 180 Ringing | m=audio 49172 RTP/AVP 0
|<-----------------------| a=rtpmap:0 PCMU/8000
| |
| 200 OK |
|<-----------------------| SIP/2.0 200 OK
| ACK | m=audio 3456 RTP/AVP 0
|----------------------->| a=rtpmap:0 PCMU/8000
| Both Way RTP Media |
|<======================>|
| |
| BYE |
|<-----------------------|
| 200 OK |
|----------------------->|
| |
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 Message syntax (I)
Request Line
Message
or Payload
Headers
Status Line

• SIP is request / response

• Every request carries a method indicating
the request type
INVITE sip:bob@[Link] SIP/2.0
• Every response carries a status code
explaining the answer
SIP/2.0 200 OK

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 Message syntax (II)
• Both carry message headers with the
parameters
– Variable depending on method / context
– Additional information about the message
– Ex: From: Alice <sip:alice@[Link]>

• Payload transports additional info (normally

using SDP)
– No user data!
– description of audio / video capabilities
– Ex: m=audio 3456 RTP/AVP 0
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 SIP addressing
• SIP URL (Uniform Resource Locator)
– Specify the location of a resource (user)
– Email-like addresses
– sip:User@Domain
– But it is not an e-mail address!
• sip:alice@[Link]
• sip:33445566@[Link]
• Non sip URLs can be also found in SIP
messages
– mailto, tel. im …

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 Original SIP methods (I)
• Six methods defined initially in the
standard
– INVITE
– ACK
– BYE
– CANCEL
– OPTIONS
– REGISTER

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 Original SIP methods (II)
• INVITE initiates sessions
– session description included in message body
– re-INVITEs used to change session state

• ACK confirms session establishment

– can only be used with INVITE

• BYE terminates sessions

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 Original SIP methods (III)
• CANCEL cancels a pending INVITE

• OPTIONS capability inquiry

• REGISTER binds a permanent address to

current location
– may convey user data (CPL scripts)

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 SIP methods extensions
• INFO mid-call signaling
• COMET preconditions met
• PRACK provisional reliable responses ACK
• SUBSCRIBE/NOTIFY/MESSAGE instant messaging
• REFER call re-direction
• UPDATE update call state
• PUBLISH user info

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 SIP responses (I)
• Borrowed from HTTP:
 xyz code + explanatory text
• Receivers need to understand x
• 1yz Informational
– 100 Trying
– 180 Ringing
– 183 Session in progress
• 2yz Success
– 200 ok

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 SIP responses (II)
• 3yz Redirection
– 300 Multiple choices
– 302 Moved temporarily
– 305 Use proxy

• 4yz Request failure

– 400 Bad Request
– 401 Unauthorized
– 407 Proxy authentication required
– 486 Busy Here
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 SIP responses (III)
• 5yz Server failure
– 500 Server internal error
– 501 Not implemented
– 503 Service unavailable

• 6yz Global failure

– 600 Busy everywhere
– 606 Not acceptable

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 A basic SIP call (again)
Alice Bob
| | INVITE sip:bob@[Link] SIP/2.0
| INVITE | From: Alice <sip:alice@[Link]>
|----------------------->| To: Bob <sip:bob@[Link]>
| 180 Ringing | m=audio 49172 RTP/AVP 0
|<-----------------------| a=rtpmap:0 PCMU/8000
| |
| 200 OK |
|<-----------------------| SIP/2.0 200 OK
| ACK | m=audio 3456 RTP/AVP 0
|----------------------->| a=rtpmap:0 PCMU/8000
| Both Way RTP Media |
|<======================>|
| |
| BYE |
|<-----------------------|
| 200 OK |
|----------------------->|
| |
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 SIP message headers
• Huge number of different headers
– See RFC 3261
– Nomenclature: SIP header = Other protocol's
header field
• Headers parameterize the request or
response
• A SIP entity ignores headers it does not
understand
– Simplifies interworking

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 Header types
• General headers
– Basic info to process the transaction
– E.g. “To”, “From” and “Call-ID” headers
• Request headers
– Additional info about request or sender
– E.g. The “Subject” header
• Response headers
– Additional info about response or receiver
– E.g. “Unsupported” and “Retry-After” header
• Entity headers
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A complete SIP message

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Example: Invite message

Starting Line

Headers

Separator

Message Body

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 SIP entities: UAC & UAS
• User Agent Client (UAC)
– Places calls
– Sends requests

• User Agent Server (UAS)

– Waits for incoming calls
– Sends responses

• A SIP terminal is a UAC + UAS

– Beware of this special client/server terminology!
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 SIP network entities
• Servers (UAS)
– Proxy server
– redirect server
– registrar
• Can be combined in a single device
 p.e. SER!!!
• They deal only with signaling
• Media is transported end-to-end
– There are exceptions...

 Signaling controls media interchange: It is not

necessary to touch the media streams directly!
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 SIP entities: Registrar
• Users register their contact info
– Keeps the info at a Location Server
• (Which is simply a database)
– SIP address, but also other addresses, with a type
indication:
sip:[Link]@[Link]
sip:935422942@[Link], user = phone
– Enables personal and terminal mobility and user
behaviour patterns

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Registration example

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 Registration example
• Contact could also be a specific location
e.x. [Link]

• Un-Register
– Send the same message but with
EXPIRE = 0

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 SIP entities: Proxy server
• Proxy Server
– Acts on behalf of UAC
• Processing the request
• Responding to requests
• Forwarding requests
– Can impersonate the client
– Acts both as client and server
– Forking proxies
• Forwards an INVITE to several destinations in
parallel
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Proxy example

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 Stateless vs Stateful proxy
• In general
– Stateless: get a request, forward it, forget about
it
– Stateful: remembers previous responses/requests
of a particular transaction
act more intelligently on subsequent requests!

• Record-Route: force future messages to pass

from this proxy

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 SIP entites: Redirect server
• Maps destination SIP addresses to
“reachable” addresses
– SIP address, e-mail, telephone number,
etc.
– Depending on user preferences
• Informs the client about them
– No message forwarding!

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Redirection example

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 Additional examples
(taken from RFC 3665 and Collins)

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 Session establishment
through two proxies
In this scenario, Alice completes a call to Bob using two proxies Proxy 1
and Proxy 2.
The initial INVITE (F1) contains a pre-loaded Route header with the
address of Proxy 1 (Proxy 1 is configured as a default outbound proxy
for Alice). The request does not contain the Authorization credentials
Proxy 1 requires, so a 407 Proxy Authorization response is sent
containing the challenge information.
A new INVITE (F4) is then sent containing the correct credentials and the
call proceeds.
The call terminates when Bob disconnects by initiating a BYE message.
Proxy 1 inserts a Record-Route header into the INVITE message to
ensure that it is present in all subsequent message exchanges. Proxy
2 also inserts itself into the Record-Route header.
The ACK (F15) and BYE (F18) both have a Route header.

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 Unsuccessful temporarily
unavailable
In this scenario, Bob initially sends a 180 Ringing
response to Alice, indicating that alerting is taking
place.
However, then a 480 Unavailable is then sent to Alice.
This response is acknowledged and then proxied back
to Alice.

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Alice Proxy 1 Proxy 2 Bob
| | | |
| INVITE F1 | | |
|--------------->| INVITE F2 | |
| 100 Trying F3|--------------->| INVITE F4 |
|<---------------| 100 F5 |--------------->|
| |<---------------| 180 F6 |
| | 180 F7 |<---------------|
| 180 Ringing F8|<---------------| |
|<---------------| | 480 F9 |
| | |<---------------|
| | | ACK F10 |
| | 480 F11 |--------------->|
| |<---------------| |
| | ACK F12 | |
|480 Temporarily |--------------->| |
| Unavailable F13| | |
|<---------------| | |
| ACK F14 | | |
|--------------->| | |
| | | |
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 Forking proxy
• Used if a particular user is registered at
several locations
• Stateful!
– Proxy remembers that it has two sessions
pending and it cancels the one the moment the
other responds

• The “branch” parameter in the “via” header

allows to distinguish among forked requests
(and responses)

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 SDP
• Session Description Protocol
– Which is not a protocol...
– ...but a description language for media flows

• Convey sufficient information to participate

in a multimedia session

• Independent of SIP
– But they work nicely together

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 SDP message info
• SDP includes description of:
– Media to use
• Codec
• Sampling rate
– Media destination
• IP address and port number
– Session name and purpose
– Times the session is active
– Contact information

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 SDP message example

v=0
o=alice 28908044538 289080890 IN IP4 [Link]
s=Wedding Proposal
e=alice@[Link]
c=IN IP4 [Link]
t=28908044900 28908045000
m=audio 49170 RTP/AVP 15 98
a=rtpmap:15 G728/8000
a=rtpmap:98 L16/11025/2
m=audio 4666 RTP/AVP 4
a=rtpmap:4 G723/8000

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 SDP capability negotiation
• Caller offers set of possible media
• Callee accepts or rejects every one of them
– If several chosen, re-invite
• If none acceptable, warning sent back and
re-invite
• Alternative: Use the OPTIONS method
– Query the callee's capabilities prior to INVITE

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 SDP examples
[Link] capability selection with SDP
[Link]-invite after capability exchange with SDP
[Link] query with OPTIONS

(Taken from Collins)

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 Example 1:

Successful
capability
selection

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 Example 2:

Re-Invite after
capability
exchange

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 Example 3:

capability
query with
OPTIONS

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 For next class
Find an article on any of the following topics:
- SIP + WLANs
- Mobile P2PSIP
- Application layer mobility using SIP

Small presentation (5-10 slides) each during

seminar
Short report (max. 2 pages) on paper send by
Friday, 29 January

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