Anna Sfairopoulou Page 1
SIP
Anna Sfairopoulou Page 2
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
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Why SIP
• SIP VoIP + WLAN competition to cellular telephony?
• IMS component• P2PSIP• Application layer mobility using SIP
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The si
mplest voice call
10.0.0.1192.168.20.2
Listening on port 5060Expecting UDPExpecting CODEC PCM A-law
Knows port 5060Knows UDPKnows CODEC PCM A-law
Knows address 192.168.20.2
Internet
Alice
Bob
Self-written C code:•Open socket•Take voice samples from microphone•Send voice samples in UDP packets
Self-written Java code:•Open socket•Wait for UDP packets•Playback voice samples
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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– TCP
• To control that interchange, signaling protocols are used– RSVP– SIP– RTSP
(+ RTP on top)
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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 “si
mplest” voice call
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The real “si
mplest” voice call
SIP Express
Server (SER)
Database (MySQL)
DNS (Named)
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
SIP
TCP / UDP / SCTP?
MPEG, H.261, etc.
RTP
UDP
IP
RTCP, TRIP, etc.
TCP / UDP?
Fast/GigEth, ATM, 802.11 MAC/LLC, etc.
DSL, Phy GigEth,802.11 PHY etc.
Signaling Media Transport Support
Physical
Network
Transport
Session
Presentation
Layer
Datalink
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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 | |----------------------->| | 180 Ringing | |<-----------------------| | | | 200 OK | |<-----------------------| | ACK | |----------------------->| | Both Way RTP Media | |<======================>| | | | BYE | |<-----------------------| | 200 OK | |----------------------->| | |
INVITE sip:[email protected] SIP/2.0From: Alice <sip:[email protected]>To: Bob <sip:[email protected]>m=audio 49172 RTP/AVP 0a=rtpmap:0 PCMU/8000
SIP/2.0 200 OKm=audio 3456 RTP/AVP 0a=rtpmap:0 PCMU/8000
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Message syntax (I)
• SIP is request / response• Every request carries a method indicating
the request typeINVITE sip:[email protected] SIP/2.0
• Every response carries a status code explaining the answer
SIP/2.0 200 OK
Request Line
or
Status Line
Message
HeadersPayload
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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:[email protected]>
• 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:[email protected]• sip:[email protected]
• 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 | |----------------------->| | 180 Ringing | |<-----------------------| | | | 200 OK | |<-----------------------| | ACK | |----------------------->| | Both Way RTP Media | |<======================>| | | | BYE | |<-----------------------| | 200 OK | |----------------------->| | |
INVITE sip:[email protected] SIP/2.0From: Alice <sip:[email protected]>To: Bob <sip:[email protected]>m=audio 49172 RTP/AVP 0a=rtpmap:0 PCMU/8000
SIP/2.0 200 OKm=audio 3456 RTP/AVP 0a=rtpmap:0 PCMU/8000
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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:[email protected]:[email protected], 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 locatione.x. pc1.grup2.labtel2.upf.edu
• 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 193.175.132.118
s=Wedding Proposal
c=IN IP4 126.16.69.4
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
1.Successful capability selection with SDP2.Re-invite after capability exchange with SDP3.Capability 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