hgs/SIP Tutorial 1

The Session Initiation Protocol

(SIP)
Henning Schulzrinne
Dept. of Computer Science
Columbia University
New York, New York
(sip:)[email protected]

May 2001
hgs/SIP Tutorial 2

Overview

• protocol architecture
• typical component architectures
• addressing and locating SIP entities
• protocol operation and extensions
• reliability
• services, features and caller preferences
• security and QoS
• programming SIP services

May 2001
hgs/SIP Tutorial 3

Introduction

• SIP = core protocol for establishing sessions in the Internet

• transports session description information from initiator (caller) to callees
• allows to change parameters in mid-session
• terminate session

May 2001
hgs/SIP Tutorial 4

VoIP protocol architecture

Languages/APIs
JAIN CPL voiceXML
Parlay servlets
sip−cgi

Directory/Discovery peer−to−peer Signaling

DNS/enum LDAP TRIP SIP SDP MGCP RTSP
PINT H.248
SLP SPIRITS
master−slave

QoS Transport
DiffServ IntServ RTP TLS
SCTP

May 2001
hgs/SIP Tutorial 5

Multimedia protocol stack

signaling quality of service media transport

MGCP/Megaco

SDP reservation measurement media encaps.

(H.261, MPEG)

application
H.323 SIP RTSP RSVP RTCP

daemon
RTP
transport

TCP UDP
network

IPv4, IPv6

kernel
PPP AAL3/4 AAL5 PPP
link

Sonet ATM Ethernet V.34

physical

May 2001
hgs/SIP Tutorial 6

SIP protocol use

LDAP address TRIP

lookup

PSTN gateway lookup

DNS next−hop

SIP SIP
signaling SDP SDP
may sets up
trigger
RSVP RSVP
media RTP RTP
UDP UDP

May 2001
hgs/SIP Tutorial 7

SIP applications

• setting up voice-over-IP calls

• setting up multimedia conferences
• event notification (subscribe/notify) ➠ IM and presence
• text and general messaging
• signaling transport

May 2001
hgs/SIP Tutorial 8

SIP addressing

May 2001
hgs/SIP Tutorial 9

Personal mobility

SIP uses email-style addresses to identify users

[email protected]
(also used by [email protected])

yahoo.com

tel:12128541111
[email protected]

[email protected]

[email protected]

columbia.edu tel:12015551234
[email protected]

[email protected]

May 2001
hgs/SIP Tutorial 10

SIP addressing

• typically, same as user’s email address:

[email protected]
[email protected]
• written as URL, e.g., sip:[email protected]
• can add parameters, such as type (user=”phone”) or transport protocol

May 2001
hgs/SIP Tutorial 11

tel URLs (RFC 2806)

• also can use tel URLs for telephone numbers, e.g., tel:+12125551212 or
fax:+358.555.1234567
• either global (tel:+1...) or local
(tel:0w003585551234567;phone-context=+3585551234 numbers
• allow post-dialing digits: ;postd=pp32
• also modem:+3585551234567;type=v32b?7e1;type=v110

May 2001
hgs/SIP Tutorial 12

SIP building blocks

SIP user agent IP phone, PC, conference bridge

SIP redirect server returns new location for requests

SIP stateless proxy routes call requests

SIP (forking) proxy routes call requests

A@
B@
C@
SIP registrar maintains mappings from names to addresses

May 2001
hgs/SIP Tutorial 13

Back-to-back UA (B2BUA)

• two (or more) user agents, where incoming calls trigger outgoing calls to
somebody else
• also, “third-party call control” (later)
• useful for services and anonymity

SIP UA1 (UAS) SIP UA2 (UAC)

INVITE b2b INVITE callee

200 OK
200 OK

May 2001
hgs/SIP Tutorial 14

Maintaining state in SIP entities

Stateless: each request and response handled indepdently

(Transaction) stateful: remember a whole request/response transaction
Call stateful: remember a call from beginning to end

May 2001
hgs/SIP Tutorial 15

SIP building block properties

media stateless stateful call state

UA (UAC, UAS) yes no unlikely common
proxy no yes common possible (firewall)
redirect registrar no no yes N/A

May 2001
hgs/SIP Tutorial 16

SIP architecture: peer-to-peer

SIP
redirect server

user agent (UA)

user agent (UA) user agent (UA)

128.119.40.186
Internet

RTP audio

128.59.19.141

CATV
Ethernet

May 2001
hgs/SIP Tutorial 17

SIP architecture: outbound proxy

wonderland.com macrosoft.com
REGISTER sip:macrosoft.com SIP/2.0
To: sip:[email protected]
From: sip:[email protected]
Contact: sip:[email protected]

outbound registrar

proxy Internet proxy

wonderland.com
INVITE sip:[email protected] SIP/2.0 INVITE sip:[email protected] SIP/2.0
[email protected]
[email protected]
INVITE sip:[email protected] SIP/2.0

May 2001
hgs/SIP Tutorial 18

SIP architecture: VoIP to PSTN

location server

TRIP

SLP?, TRIP−GW?

sip:[email protected]
sip:1−212−555−1234@domain
SIP

H.248
tel:+1−212−555−1234

outbound proxy
IP
010

May 2001
hgs/SIP Tutorial 19

SIP architecture: PSTN to VoIP

enum database
DNS
4.3.2.1.5.5.5.2.1.2.1.e164.arpa
SCP enum

sip:[email protected]

INVITE sip:[email protected]
IP
010

May 2001
hgs/SIP Tutorial 20

SIP operation in proxy mode

cs.columbia.edu

? location server
1
4
INVITE
cs.tu−berlin.de INVITE hgs@play 5

hgs@play
henning
[email protected] 2
3
200 OK
200 OK
[email protected]−berlin.de 6
7 play

tune
8

ACK hgs@play

9 media stream

May 2001
hgs/SIP Tutorial 21

SIP operation in redirect mode

ieee.org
? location
server

columbia.edu
1
tu-berlin.de INVITE [email protected] 2

henning
3

302 Moved temporarily

Contact: [email protected]
4

5 ACK [email protected]

columbia.edu
6 INVITE [email protected]

7 200 OK

8 ACK [email protected] hgs

(302: redirection for single call; 301 permanently)

May 2001
hgs/SIP Tutorial 22

Locating SIP users

May 2001
hgs/SIP Tutorial 23

Locating users: registrars and location servers

registrar example.com
REGISTER
[email protected] A@ SQL, LDAP, Corba,
B@ proprietary, ...
Contact: C@
alice@pc17

location server
INVITE
[email protected] INVITE [email protected]

proxy

May 2001
hgs/SIP Tutorial 24

Basic user location mechanism

1. host(SIP URL) −→ host name of proxy

2. DNS: host name of proxy → SIP server(s)
3. if SIP UAS: alert user; done
4. if SIP proxy/redirect server: map URLn −→ URLn+1 , using any information in
request
5. go to step 1
One minor exception. . .

May 2001
hgs/SIP Tutorial 25

Basic SIP “routing” mechanisms

• will fill in details later

• route using request URIs
• all but first request in call typically bypass proxies and go direct UAC – UAS
• however, can use “record-routing” to force certain proxies to be visited all the time
• responses always traverse the same route as requests

May 2001
hgs/SIP Tutorial 26

Outbound proxies

• normally, proxy serves one or more domains

• outbound proxies are used for all outbound requests from within a domain
• typically, for managing corporate firewalls and policy enforcement
• may also provide dial plans or route tel/fax URLs
• other uses: lawyer client billing, . . .

May 2001
hgs/SIP Tutorial 27

Locating users: DNS SRV

• email: DNS MX record allows mapping of domain to mail host, e.g.

host -t mx yahoo.com
yahoo.com MX 1 mx2.mail.yahoo.com
yahoo.com MX 1 mx3.mail.yahoo.com
yahoo.com MX 1 mx1.mail.yahoo.com
yahoo.com MX 9 mta-v1.mail.yahoo.com
• SIP: use a newer record for general-purpose mapping, SRV (RFC 2782)
• mapping from service and transport protocol to one or more servers, including
protocols
_sip._tcp SRV 0 0 5060 sip-server.cs.columbia.edu.
SRV 1 0 5060 backup.ip-provider.net.
_sip._udp SRV 0 0 5060 sip-server.cs.columbia.edu.
SRV 1 0 5060 backup.ip-provider.net.
• allows priority (for back-up) and weight (for load balancing)

May 2001
hgs/SIP Tutorial 28

Using DNS SRV for scalable load-balancing

a.example.com
_sip._udp SRV 0 0 a1.example.com
SRV 1 0 a2.example.com
a1.example.com, a2.example.com

a*@example.com

s1.example.com

s2.example.com b*@example.com
sip:[email protected] sip:[email protected]

s3.example.com

_sip._udp SRV 0 0 s1.example.com b1.example.com, b2.example.com

SRV 0 0 s2.example.com
SRV 0 0 s3.example.com

May 2001
hgs/SIP Tutorial 29

Aside: SIP scaling

• HTTP request director ↔ SIP client-based

• HTTP randomized DNS (short TTL!) ↔ SRV weights and priorities
• can’t just distribute requests randomly, since backend (registration)
synchronization is needed
• registration scaling: requests/second * 3600; e.g., 100 requests/second ➠ 360,000
users/server
• major bottlenecks are logging and database updates
• generally, higher registration than INVITE rates

May 2001
hgs/SIP Tutorial 30

SIP protocol operation

May 2001
hgs/SIP Tutorial 31

SIP requests and responses

• text, not binary, format

• look very similar to HTTP/1.1
• requests and responses are similar except for first line
• requests and responses can contain message bodies: typically session descriptions,
but also ASCII or HTML

May 2001
hgs/SIP Tutorial 32

SIP syntax

request response

method URL SIP/2.0 SIP/2.0 status reason

Via: SIP/2.0/ protocol host:port
From: user <sip:from_user@source>

message header
To: user <sip:to_user@destination>
Call−ID: localid@host
CSeq: seq# method
Content−Length: length of body
Content−Type: media type of body
Header: parameter ;par1=value ;par2="value"
;par3="value folded into next line"
blank line

message body
V=0
o= origin_user timestamp timestamp IN IP4 host
c=IN IP4 media destination address
t=0 0
m= media type port RTP/AVP payload types

message

May 2001
hgs/SIP Tutorial 33

SIP syntax

• field names and some tokens (e.g., media type) are case-insensitive
• everything else is case-sensitive
• white space doesn’t matter except in first line
• lines can be folded
• multi-valued header fields can be combined as a comma-list

May 2001
hgs/SIP Tutorial 34

SIP methods

INVITE initiate call

ACK confirm final response
BYE terminate (and transfer) call
CANCEL cancel searches and “ringing”
OPTIONS features support by other side
REGISTER register with location service
INFO mid-call information (ISUP)
COMET precondition met
PRACK provisional acknowledgement
SUBSCRIBE subscribe to event
NOTIFY notify subscribers
REFER ask recipient to issue SIP request (call transfer)

May 2001
hgs/SIP Tutorial 35

SIP invitation and media negotiation

[email protected] calls [email protected]

INVITE sip:[email protected] SIP/2.0 SIP/2.0 200 OK

From: sip:[email protected] From: sip:[email protected]
To: sip:[email protected] To: sip:[email protected]
Call−ID: [email protected] Call−ID: [email protected]
CSeq: 42 INVITE CSeq: 42 INVITE
Content−Type: application/sdp Content−Type: application/sdp
v=0 v=0
o=user1 536 2337 IN IP4 h3.wonderland.com o=user1 535 687637 IN IP4 m.macrosoft.com
c=IN IP4 h3.wonderland.com c=IN IP4 m.macrosoft.com
m=audio 3456 RTP/AVP 0 1 m=audio 1200 RTP/AVP 1
m=video 4000 RTP/AVP 38 39 m=video 0 RTP/AVP

accept audio, decline video

May 2001
hgs/SIP Tutorial 36

Tagging To

• after forking and merging, hard to tell who responded

• UAS responds with random tag added to disambiguate
To: "A. G. Bell" <sip:[email protected]>
;tag=a48s
• future requests are ignored if they contain the wrong tag

May 2001
hgs/SIP Tutorial 37

SIP call legs

• call leg: From, To, Call-ID

• requests from callee to caller reverse To and From
• caller and callee keep their own CSeq space
• either side can send more INVITEs or BYE

May 2001
hgs/SIP Tutorial 38

SIP responses

Informational Success Redirection Request Failure

100 Trying 200 OK 300 Multiple Choices 400 Bad Request

180 Ringing 301 Moved Perm. 401 Unauthorized
181 Call forwarded 302 Moved Temp. 403 Forbidden
182 Queued 380 Alternative Serv. 404 Not Found
183 Session Progress 405 Bad Method
415 Unsupp. Content
420 Bad Extensions
486 Busy Here

500 Server Error 600 Busy Everwhere

501 Not Implemented 603 Decline
503 Unavailable 604 Doesn’t Exist
504 Timeout 606 Not Acceptable

Server Failure Global Failure

May 2001
hgs/SIP Tutorial 39

SIP response routing

• requests are routed via URL

• response traces back request route without proxy server state
• forward to host, port in next Via
• TCP: re-use connection if possible, create new one if needed
• UDP: may send responses to same port as requests
Via: SIP/2.0/UDP server.domain.org:5060
;received=128.1.2.3

May 2001
hgs/SIP Tutorial 40

SIP response routing

[email protected] [email protected] [email protected]

Via: y1.yahoo.com
Via: a.example.com Via: a.example.com

INvITE

Via: a.example.com
Via: y1.yahoo.com
Via: a.example.com

Via: sip.columbia.edu
Via: y1.yahoo.com
Via: a.example.com
Via: sip.columbia.edu
Via: y1.yahoo.com
Via: a.example.com
Via: cs.columbia.edu
Via: sip.columbia.edu
Via: y1.yahoo.com
Via: a.example.com

[email protected]

200 OK

Via: cs.columbia.edu
Via: sip.columbia.edu
Via: y1.yahoo.com
Via: a.example.com

[email protected]

May 2001
hgs/SIP Tutorial 41

SIP spirals

INVITE sip:[email protected] SIP/2.0

Via: sales.acme.com;branch= h(info,alice,17,1,kelly@sales)
Via: acme.com;branch= h(info,alice,17,1,kelly@acme)
Via: sales.acme.com;branch= h(info,alice,17,1,bob@sales)
Via: acme.com;branch= h(info,alice,17,1,info)
Via: ph123.wonderland.com

INVITE sip:[email protected] SIP/2.0

Via: acme.com;branch= h(info,alice,17,1,info)
Via: ph123.wonderland.com
INVITE sip:[email protected] SIP/2.0
To: sip:[email protected]
Via: ph123.wonderland.com
acme.com sales.acme.com

INVITE sip:[email protected] SIP/2.0

Via: acme.com;branch=
h(info,alice,17,1,kelly@acme)
Via: sales.acme.com;branch= h(info,alice,17,1,bob@sales)
Via: acme.com;branch= h(info,alice,17,1,info)
Via: ph123.wonderland.com
[email protected]

INVITE sip:[email protected] SIP/2.0

Via: sales.acme.com;branch=
h(info,alice,17,1,bob@sales)
Via: acme.com;branch= h(info,alice,17,1,bob@sales)
branch= h(To,From,Call−ID,CSeq,URL) Via: ph123.wonderland.com

May 2001
hgs/SIP Tutorial 42

Forcing request paths

• usually, bypass proxies on subsequent requests

• some proxies want to stay in the path → call-stateful:
– firewalls
– anonymizer proxies
– proxies controlling PSTN gateways
• use Record-Route and Route

May 2001
hgs/SIP Tutorial 43

Request routing

A PA PB B

INVITE PB INVITE PB INVITE B

Contact: A Contact: A Contact: A
Record−Route: PA Record−Route: PB, PA

200 OK 200 OK 200 OK

Contact: B Contact: B Contact: B
Record−Route: PB, PA Record−Route: PB, PA Record−Route: PB, PA

ACK PA ACK PB ACK B

Route: PB,B Route: B

BYE A BYE A BYE A

Route: A;maddr=PB Route: A;maddr=PB
Route: A;maddr=PA
Route: A;maddr=A

May 2001
hgs/SIP Tutorial 44

SIP request forking

macrosoft.com

[email protected]

a.wonderland.com INVITE bob@b

INVITE [email protected]
CANCEL bob@c

INVITE carol@c
[email protected]

ACK 200 OK

BYE [email protected]

200 OK

May 2001
hgs/SIP Tutorial 45

SIP sequential request forking

Use q values to govern order of sequential search:

q=1.0

1 302 Moved temporarily

2 q=0.7
INVITE
486 Busy here

3
q=0.1
200 OK

May 2001
hgs/SIP Tutorial 46

SIP request forking

• branches tried in sequence or parallel (or some combination)

• recursion: may try new branches if branch returns 3xx
• return best final answer = lowest status code
• forward provisional responses

May 2001
hgs/SIP Tutorial 47

Parallel forking call flow

alice bob@portal bob@home bob@work

INVITE bob@portal

100 Trying

180 Ringing

200 OK
Contact: bob@home

200 OK
CANCEL bob@work

200 OK (CANCEL)

487 Cancelled (INVITE)

ACK bob@work

ACK bob@home

UAC proxy server UAS UAS

May 2001
hgs/SIP Tutorial 48

SIP transport issues

• SIP operates over any packet network, reliable or unreliable

• choices:
UDP: most common
– low state overhead
– small max. packet size
TCP: can combine multiple signaling flows over one link
– use with SSL
– connection setup overhead
– HOL blocking for trunks
SCTP: new protocol
– no HOL blocking
– fallback address (but SRV provides this already)
– connection setup overhead

May 2001
hgs/SIP Tutorial 49

Transport reliability for all but INVITE

client UAS, proxy

BYE
500 ms

• used for BYE, OPTIONS,

1s
SUBSCRIBE, NOTIFY, . . .
• 1xx sent by UAS or proxy 2s
4s
only if no final answer ex- 4s
...
pected within 200 ms no more than
11 packets
• if provisional response, re-
transmit with T 2 (4) seconds 200, 4xx, 5xx, 6xx

May 2001
hgs/SIP Tutorial 50

INVITE reliability

• INVITE is special – long time

between request and final re- Initial
sponse −
INVITE
• 100 (by proxy) indicates re- T1*2
n
Calling
INVITE
quest has been received
1xx
• proxy usually forwards 1xx
status 1xx 7 INVITE sent
from all branches ACK
Call proceeding

• only retransmit until 100

status
ACK

• ACK confirms receipt of final status

ACK
Completed
response event
request sent

May 2001
hgs/SIP Tutorial 51

Other signaling approaches

May 2001
hgs/SIP Tutorial 52

Differences to classical signaling

name examples network “channel”

in-band E&M, DTMF same same
out-of-band ISUP, Q.931 different different
IP SIP typically same different
IP signaling meets media only at end systems, while PSTN out-of-band intersects at
every switch

May 2001
hgs/SIP Tutorial 53

Aside: Alternative architecture: master-slave

• master-slave: MGC (media gateway controller) controls one or more gateways

• allows splitting of signaling and media functionality
• “please send audio from circuit 42 to 10.1.2.3”
• uses MGCP (implemented) or Megaco/H.248 (standardized, but just beginning to
be implemented)
• gateway can be residential
• basis of PacketCable NCS (network control system) architecture
• service creation similar to digital PBX or switch
• end system has no semantic knowledge of what’s happening
• −→ can charge for caller id, call waiting

May 2001
hgs/SIP Tutorial 54

MGCP/SIP architecture

STP

call agent SIP call agent SIP TCAP

MG controller MG controller SS7 gwy SCP
H.323 H.323 ISUP

MGCP/Megaco MGCP/Megaco SS7

RGW

TGW PSTN
Internet
RTP
RGW

May 2001
hgs/SIP Tutorial 55

Extending SIP

extension behavior determine?

new headers ignored –
new headers mandatory Supported
new method OPTIONS
new body type Accept
new status code class-based
new URL type ?

May 2001
hgs/SIP Tutorial 56

SIP extensions and feature negotiation

• if crucial, mark with “Require: feature”

• IANA-registered features are simple names, private features use reverse domain
names
• indicate features supported in Supported:
C->S: INVITE sip:[email protected] SIP/2.0
Require: com.example.billing
Supported: 100rel
Payment: sheep_skins, conch_shells

S->C: SIP/2.0 420 Bad Extension

Unsupported: com.example.billing

S->C: SIP/2.0 421 Extension Required

Require: 183

May 2001
hgs/SIP Tutorial 57

User identification

May 2001
hgs/SIP Tutorial 58

Standard call/caller identification

Request-URI: next hop

To: logical call destination
From: logical call origin
Organization: organization of caller/callee
Subject: subject of call
Call-Info: additional information about caller or callee
Call-Info:
<http://wwww.example.com/alice/photo.jpg> ;purpose=icon,
<http://www.example.com/alice/> ;purpose=info

User-Agent: make and model of user agent

May 2001
hgs/SIP Tutorial 59

Additional call information

Priority: call priority: emergency, urgent, normal, non-urgent

Alert-Info: render instead of ring tone
Alert-Info: <http://wwww.example.com/sounds/moo.wav>
In-Reply-To: call-id being returned

May 2001
hgs/SIP Tutorial 60

draft-ietf-sip-privacy

• To/headerFrom are chosen by end system ➠ may lie

• need privacy indications similar to caller id
Remote-Party-ID: "John Doe"
<sip:[email protected]>;party=calling;
id-type=subscriber;privacy=full
• screen=yes: was verified by proxy
• type can be subscriber, user, alias, return (calls), term (terminal)
• may add geographic user location

May 2001
hgs/SIP Tutorial 61

SIP services

May 2001
hgs/SIP Tutorial 62

Invitation modes

signaling media
unicast multicast
unicast telephony multicast session
multicast reach first dept. conference
➠ SIP for all modes, SAP/SDP also for multicast/multicast

May 2001
hgs/SIP Tutorial 63

SIP-based services

Call forwarding: basic INVITE behavior (proxy/redirect)

Call transfer: REFER method (see later)
Call hold: set media address to 0.0.0.0 – can be done individually per media
Caller id: From, plus extensions
DTMF carriage: carry as RTP payload (RFC 2833)
Calling card: B2BUA + voice server
Voice mail: UA with special URL(s) + possibly RTSP

May 2001
hgs/SIP Tutorial 64

Call transfer

B1

3 BYE A

1 REFER B2
Referred−By: B1
A

2 INVITE B2
Referred−By: B1

B2

May 2001
hgs/SIP Tutorial 65

IVR and VoiceXML

A@
B@
C@ SQL, LDAP

SIP SIP UA
REFER VoiceXML
200
VoiceXML
scripts
text
RTP

text

May 2001
hgs/SIP Tutorial 66

Third-party call control

RTP

200
200 3
INVITE
2 SDP (from 2) 6
5
ACK
SDP (from 4) 4

INVITE
no SDP 1
SIP ACK

SIP

May 2001
hgs/SIP Tutorial 67

SIP billing/charging

What for? How?

• transport ➠ resource reservation • resource reservation protocols
protocol • SIP-in-DIAMETER approach
• SIP services (call processing) ➠ • server log files
authentication
• PSTN gateway services
• media server services (translation,
storage)

May 2001
hgs/SIP Tutorial 68

Security issues

May 2001
hgs/SIP Tutorial 69

Threats

• spoofing From in REGISTER: call redirection

• spoofing From in INVITE: bypass call filtering
• snooping media packets
• billing confusion (identifier munging)
• denial-of-service attacks

May 2001
hgs/SIP Tutorial 70

SIP security

layer/mechanism approach characteristics

network layer IPsec adjacent nodes, all or nothing, hard to configure
transport layer TLS adjacent nodes, all or nothing
SIP INVITE basic/digest shared secrets with random parties
SIP REGISTER basic/digest securing headers?
SIP general S/MIME in progress
Basic (plaintext password) and digest (challenge-response) are very similar to HTTP
security mechanisms.

May 2001
hgs/SIP Tutorial 71

SIP authentication

Basic: include plain-text password in request, immediately or after 401

(Unauthorized) or 407 (Proxy Authorization) response
Digest: challenge-response with shared secret
Certificate: sign non-Via parts of request headers, body with PGP, PKCS #7
SSL, SSH: but only for TCP
• but: need more elaborate cryptographic capability indication in SDP

May 2001
hgs/SIP Tutorial 72

Basic authentication

• Challenge by UAS:
SIP/2.0 401 Unauthorized
WWW-Authenticate: Basic realm="business"
• client responds with
INVITE sip:[email protected] SIP/2.0
CSeq: 2 INVITE
Authorization: QWxhZGRpbjpvcGVuIHNlc2FtZQ==
where authorization is base64(userid:password)
• usually caller → callee, but challenge can be in request

May 2001
hgs/SIP Tutorial 73

Digest authentication

• A calls B and fails:

SIP/2.0 401 Unauthorized
Authenticate: Digest realm="GW service",
domain="wcom.com",
nonce="wf84f1ceczx41ae6cbe5aea9c8e88d359",
opaque="42", stale="FALSE", algorithm="MD5"
• A tries again:
INVITE sip:[email protected] SIP/2.0
Authorization:Digest username="UserA",
realm="GW service",
nonce="wf84f1ceczx41ae6cbe5aea9c8e88d359",
opaque="42", uri="sip:[email protected]",
response="42ce3cef44b22f50c6a6071bc8"

May 2001
hgs/SIP Tutorial 74

Digest authentication

username: user authenticating herself

realm: several per user, used also for display
nonce: copied into Authorization
opaque: copied into Authorization
uri: original request URL
response: 32 hex digits:
KD (H(A1 ), nonce-value : H(A2 ))
for MD5: H(H(A1 ) : nonce-value : H(A2 )))
where A1 = username : realm : passwd
A2 = method : uri

May 2001
hgs/SIP Tutorial 75

Quality of Service

May 2001
hgs/SIP Tutorial 76

Quality of service

• SIP and data paths disjoint ➠ SIP can’t reserve resources

• but: SDP may provide information to end systems on desired QoS
• SDP contains range of codecs to allow mid-call adaptation

May 2001
hgs/SIP Tutorial 77

Interaction with resource reservation

avoid “fast busy” after ringing ➠ interleave

INVITE [email protected]

183 Session Progress (SDP)

PRACK

200 OK (PRACK)

reservation

COMET

200 (COMET)

180 Ringing

PRACK

200 OK (PRACK)

200 OK (INVITE)

ACK (INVITE)
UAC UAS

May 2001
hgs/SIP Tutorial 78

SIP Caller Preferences

May 2001
hgs/SIP Tutorial 79

Preferences

callee: scripts, CPL, REGISTER advice in Contact, . . .

caller: help guide routing (“no home number”) and order of attempts when forking
(“try videophone first, then phone, then answering service”)
“caller proposes, callee disposes”

May 2001
hgs/SIP Tutorial 80

Extended SIP Contact header

q location preference
class business, residence
description show to caller
duplex full or half-duplex
feature call handling features
language languages spoken
media audio, video, text/numeric, . . .
mobility fixed or mobile
priority “only in case of emergency”
scheme URL schemes (tel, http, . . . )
service IP, PSTN, ISDN, pager, . . .

May 2001
hgs/SIP Tutorial 81

Contact example

q=quality gives preference.

SIP/2.0 302 Moved temporarily
Contact: sip:[email protected]
;action=redirect ;service=IP,voice-mail
;media=audio ;duplex=full ;q=0.7;
Contact: tel:+1-415-555-1212 ; service=ISDN
;mobility=fixed ;language=en,es,iw ;q=0.5
Contact: tel:+1-800-555-1212 ; service=pager
;mobility=mobile
;duplex=send-only;media=text; q=0.1; priority=urgent;
;description="For emergencies only"
Contact: mailto:[email protected]

May 2001
hgs/SIP Tutorial 82

Accept-Contact and Reject-Contact

• determine order of contacting users:

Accept-Contact: sip:[email protected] ;q=0,
;media="!video" ;q=0.1,
;mobility="fixed" ;q=0.6,
;mobility="!fixed" ;q=0.4
➠ “avoid connecting me to sales; I prefer a landline phone; try
• Reject-Contact: rule out destinations
Reject-Contact: ;class=personal

May 2001
hgs/SIP Tutorial 83

Request-Disposition

• proxy or redirect
• cancel ringing second phone after first picked up?
• allow forking?
• search recursively?
• search sequentially or in parallel?
• queue the call?
Request-Disposition: proxy, recurse, parallel

May 2001
hgs/SIP Tutorial 84

SIP presence, events and instant

messaging

May 2001
hgs/SIP Tutorial 85

SIP presence architecture

presentity
bob
macrosoft.com
presence server
PUA
PA REGISTER
alice@
example.com
SUBSCRIBE
UA PUA
NOTIFY

registrar

PUA

PA
NOTIFY

May 2001
hgs/SIP Tutorial 86

SIP presence components

Presentity: logical entity being subscribe to, e.g., [email protected], with

several agents
Registrar: receives REGISTER requests
Presence user agent (PUA): generates REGISTER, but no SUBSCRIBE or
NOTIFY ➠ any non-presence-aware SIP software
Presence agent: receive SUBSCRIBE, generate NOTIFY
Presence server: SIP proxy + PA
Presence client: SIP UA + PA

May 2001
hgs/SIP Tutorial 87

SIP presence protocol

subcriber presentity, PA
alice bob
SUBSCRIBE bob
Event: presence
From: alice
To: bob

200 OK

NOTIFY alice bob available

200 OK

NOTIFY alice bob not

available

200 OK

May 2001
hgs/SIP Tutorial 88

SIP SUBSCRIBE example

SUBSCRIBE sip:[email protected] SIP/2.0

Event: presence
To: sip:[email protected]
From: sip:[email protected]
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 1 SUBSCRIBE
Expires: 3600
Content-Length: 0

• Forked to all PUAs that have REGISTERed with method SUBSCRIBE.

• 200 (OK) response contains current state.

May 2001
hgs/SIP Tutorial 89

SIP NOTIFY example

NOTIFY sip:[email protected]
To: sip:[email protected]
From: sip:[email protected]
Call-ID: [email protected]
CSeq: 1 NOTIFY
Content-Type: application/xpidf+xml

<?xml version="1.0"?>
<!DOCTYPE presence
PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:[email protected];method="SUBSCRIBE">
<atom id="779js0a98">
<address uri="sip:[email protected];method=INVITE">
<status status="closed"/>
</address>
</atom>
</presentity>
</presence>

May 2001
hgs/SIP Tutorial 90

SIP events

• single-valued (light-switch) to complex (CD changer) to multi-valued

(temperature samples)
• both built-in and mediated (X10)
• often combined with audio/video in same system: security, industrial control,
home entertainment
• notification rates vary ➠ gradual transition to continuous media
temperature process packet
IR detector sensor control audio/video

0.01 0.1 1 10 100

events (packets) / second
• Event describes event type

May 2001
hgs/SIP Tutorial 91

Example home architecture

SUBSCRIBE [email protected]

NOTIFY [email protected]
SIP proxy
SIP user agent DO [email protected] (RGW)

INVITE [email protected]

(Work with Telcordia)

May 2001
hgs/SIP Tutorial 92

SIP IM

• send text or any other MIME type

• either as SDP-initiated session or as individual messages
• use MESSAGE

May 2001
hgs/SIP Tutorial 93

Programming SIP Services

May 2001
hgs/SIP Tutorial 94

Programming SIP services

safety language? party?

SIP-cgi same as scripting any callee
servlets same as Java Java callee
CPL very XML both
applets same as Java Java caller

May 2001
hgs/SIP Tutorial 95

Programming services

• “caller proposes, callee disposes, administrator decides”

• web = static pages −→ cgi-bin −→ Java
• “if somebody is trying to call for the 3rd time, allow mobile”
• “try office and lab in parallel, if that fails, try home”
• “allow call to mobile if I’ve talked to person before”
• “if on telemarketing list, forward to dial-a-joke”
• phone: CTI = complex, not generally for end users

May 2001
hgs/SIP Tutorial 96

cgi-bin for SIP Servers

• extend SIP user/proxy/redirect server functionality without changing server

software
• server manages retransmission, loop detection, authentication, . . .
• Perl, Tcl, VB scripts

May 2001
hgs/SIP Tutorial 97

Examples

• Call forward on busy/no answer • Third-party registration control

• Administrative screening (firewall) • Calendarbook access
• Central phone server • Client billing allocation (lawyer’s
• Intelligent user location office)
• End system busy
• Phone bank (call distribu-
tion/queueing)

May 2001
hgs/SIP Tutorial 98

cgi Script Functionality

called for any method except ACK or CANCEL

• proxying of requests
• returning responses
• generate new requests
once for each request or response or timeout

May 2001
hgs/SIP Tutorial 99

cgi Script Mechanism

environment variables: headers, methods, authenticated user, . . .

stdin: body of request
stdout: new request, meta-requests:
• CGI- requests for proxying, response, default action
• script cookie for state across messages
• reexecute on all, final response, never

May 2001
hgs/SIP Tutorial 100

Cgi Example: Call Forwarding

use DB_File;
sub fail {
my($status, $reason) = @_;
print "SIP/2.0 $status $reason\n\n";
exit 0;
}

tie %addresses, ’DB_File’, ’addresses.db’

or fail("500", "Address database failure");
$to = $ENV{’HTTP_TO’};
if (! defined( $to )) {
fail("400", "Missing Recipient");
}

May 2001
hgs/SIP Tutorial 101

$destination = $addresses{$to};

if (! defined( $destination )) {
fail("404", "No such user");
}

print "CGI-PROXY-REQUEST-TO $destination SIP/2.0\n";

print "CGI-Reexecute-On: never\n\n";
untie %addresses; # Close db file

May 2001
hgs/SIP Tutorial 102

The Call Processing Language

Jonathan Lennox
Columbia University
[email protected]

May 5, 2000

May 2001
hgs/SIP Tutorial 103

Purpose

Allow users to create simple Internet telephony services

Features:
• Creatable and editable by simple graphical tools
• Independent of signalling protocol
• Safe to run in servers

May 2001
hgs/SIP Tutorial 104

Abstract structure

busy
location proxy timeout
url: sip:jones@ timeout: 10s failure
Call String-switch example.com
field: from
match:
*@example.com
otherwise
location
url: sip:jones@
voicemail. redirect
example.com
merge: clear

May 2001
hgs/SIP Tutorial 105

Abstract structure (cont)

• Nodes and outputs — “boxes” and “arrows”

• Nodes have parameters
• Start from single root “call” node
• Progress down tree of control
• May invoke sub-actions
• Follow one output of each node, based on outcome
• Continue until we get to a node with no outputs

May 2001
hgs/SIP Tutorial 106

Textual representation

<cpl>
<subaction id="voicemail">
<location url="sip:[email protected]">
<redirect />
</location>
</subaction>

May 2001
 hgs/SIP Tutorial 107

Textual representation
<incoming>
<address-switch field="origin" subfield="host">
<address subdomain-of="example.com">
<location url="sip:[email protected]">
<proxy>
<busy> <sub ref="voicemail" /> </busy>
<noanswer> <sub ref="voicemail" /> </noanswer>
<failure> <sub ref="voicemail" /> </failure>
</proxy>
</location>
</address>
<otherwise>
<sub ref="voicemail" />
</otherwise>
</address-switch>
</incoming>
</cpl>

May 2001
hgs/SIP Tutorial 108

Textual representation

• Represent scripts as XML documents

• Incoming, outgoing scripts are separate top-level tags
• Nodes and outputs are both tags
• Parameters are tag attributes
• Multiple outputs to one input represented by subactions

May 2001
hgs/SIP Tutorial 109

Switch nodes

Switch nodes make decisions.

Structure:
<type-switch field=var>
<type condition1="value1">
action1
</type>
<type condition2="value2">
action2
</type>
<not-present>
action3
<otherwise>
action4
</otherwise>
</type-switch>

May 2001
hgs/SIP Tutorial 110

Address Switches: address

Switch based on textual strings:

is: (exact string match)
contains: substring match: only for “display”
subdomain-of: domain match: only for “host”, “tel”
Fields are “origin,” “destination,” “original-destination”, with subfields “address-type,”
“user,” “host,” “port,” “tel,” “display”

May 2001
hgs/SIP Tutorial 111

String Switches: string

Switch based on textual strings, with conditions:

is: exact string match
contain: substring match
Fields: subject, organization, user-agent

May 2001
hgs/SIP Tutorial 112

Time switches: time

Switch based on the current time at the server.

timezone: which timezone the matching should apply in
Conditions:
• year, month, date, day, timeofday
• each condition is a list of ranges: a1 − b1 , a2 − b2 , . . .
• must fall within a range of all specified conditions

May 2001
hgs/SIP Tutorial 113

Time switches: examples

<time month="12" date="25" year="1999">

December 25th, 1999, all day
<time month="5" date="4">
May 4th, every year, all day
<time day="1-5" timeofday="0900-1700">
9 AM – 5 PM, Monday through Friday, every week

May 2001
hgs/SIP Tutorial 114

Time switches: examples

<time timeofday="1310-1425,1440-1555,1610-1725"
day="2,4">
1:10 – 2:25 PM, 2:40 – 3:55 PM, and 4:10 – 5:25 PM, Tuesdays and Thursdays,
every week
<time date="1-7" day="1">
The first Monday of every month, all day

May 2001
hgs/SIP Tutorial 115

Location nodes

• A number of CPL actions (proxy, redirect) take locations

• Location nodes let you specify them
• These are full-featured nodes because we might want to make decisions based on
outcomes of location lookups, or cascade locations
• A CPL script has an implicit global list of locations
• Location nodes can add to this list, or clear the list

May 2001
hgs/SIP Tutorial 116

Simple location nodes: location

Specify a location explicitly.

url: explicitly specified location
clear: clear earlier location values
Only one output; cannot fail. Don’t use an explicit output node in the URL.

May 2001
hgs/SIP Tutorial 117

Location lookup nodes: lookup

Specify a location abstractly, by where it should be looked up.

Parameters:
source: URL (ldap, http (CGI), etc) or non-URL source (“registration”) to search for
locations
timeout: time to wait
use/ignore: • use: caller-preferences parameters to use
• ignore: caller-preferences parameters to disregard
merge:
Outputs: success, notfound, failure

May 2001
hgs/SIP Tutorial 118

Location removal nodes: remove-location

Remove locations from the location set, based on caller preferences/callee capabilities.
Has the same effect as a “Reject-Contact” header.
param: caller preference parameters to apply
value: values of parameters specified in “param”
location: caller preference location to apply

May 2001
hgs/SIP Tutorial 119

Signalling Actions: proxy

Proxy the call to the currently-specified set of locations, and automatically select one
“best” final response.
timeout: time before giving up on the proxy attempt
recurse: recurse on redirect responses to the proxy attempt?
ordering: try location in parallel, sequential, first-only
• Outputs: busy, noanswer, failure
• If the proxy attempt was successful, script terminates

May 2001
hgs/SIP Tutorial 120

Signalling Actions: redirect

Redirect the call to the currently-specified set of locations. This has no specific
parameters, and causes the script to terminate.

May 2001
hgs/SIP Tutorial 121

Signalling Actions: reject

Reject the call attempt. This causes the script to terminate.

status: “busy,” “notfound,” “reject,” or “error”, or a 4xx, 5xx, or 6xx code (for SIP).
reason: string explaining the failure.

May 2001
hgs/SIP Tutorial 122

Non-signalling action: mail

Notify a user of something through e-mail.

url: the address to contact, including any header parameters.

May 2001
hgs/SIP Tutorial 123

Non-signalling action: log

Store a record of the current call in a log.

name: the name of the log this should be stored
omment: a string explaining the log entry
Outputs: success, failure

May 2001
hgs/SIP Tutorial 124

Subactions

• XML syntax defines a tree; we want CPLs to be represented as directed acyclic

graphs.
• Subactions are defined at the top level of the script, outside other actions.
• for acyclicity, top-level actions and subactions may only call subactions which
were defined earlier in the script.
• Anywhere a node is expected, you can instead have a sub tag, with a ref
parameter which refers to a subaction’s id.

May 2001
hgs/SIP Tutorial 125

Example: Call Redirect Unconditional

<cpl>
<incoming>
<location url="sip:[email protected]">
<redirect />
</location>
</incoming>
</cpl>

May 2001
hgs/SIP Tutorial 126

Example: Call Forward Busy/No Answer

<cpl>
<subaction id="voicemail">
<location url="sip:[email protected]" >
<proxy />
</location>
</subaction>

<incoming>
<location url="sip:[email protected]">
<proxy timeout="8s">
<busy>
</busy>
<noanswer>
<sub ref="voicemail" />
</noanswer>
</proxy>
</location>
</incoming>
</cpl>

May 2001
hgs/SIP Tutorial 127

Example: Call Screening

<cpl>
<incoming>
<address-switch field="origin" subfield="user">
<address is="anonymous">
<reject status="reject"
reason="I don’t accept anonymous calls" />
</address>
</address-switch>
</incoming>
</cpl>

May 2001
hgs/SIP Tutorial 128

Example: Time-of-day Routing

<?xml version="1.0" ?>

<!DOCTYPE call SYSTEM "cpl.dtd">

<cpl>
<incoming>
<time-switch timezone="US/Eastern">
<time day="1-5" timeofday="0900-1700">
<lookup source="registration">
<success>
<proxy />
</success>
</lookup>
</time>
<otherwise>
<location url="sip:[email protected]">
<proxy />
</location>
</otherwise>
</time-switch>
</incoming>
</cpl>

May 2001
hgs/SIP Tutorial 129

Example: Non-call Actions

<?xml version="1.0" ?>

<!DOCTYPE call SYSTEM "cpl.dtd">

<cpl>
<incoming>
<lookup source="http://www.example.com/cgi-bin/locate.cgi?user=jones"
timeout="8">
<success>
<proxy />
</success>
<failure>
<mail url="mailto:[email protected]&Subject=lookup%20failed" />
</failure>
</lookup>
</incoming>
</cpl>

May 2001
hgs/SIP Tutorial 130

SIP for Third-Generation Wireless

Networks

May 2001
hgs/SIP Tutorial 131

3G networks

• successor to 2G mobile networks: GSM (TDMA) and IS-95 (CDMA) in 900/1800

MHz range
• 2.5G: GSM −→ GPRS −→ EDGE
• use different air interfaces in 2 GHz range: W-CDMA, CDMA 2000, TD-CDMA
• 3GPP standardizes for W-CDMA (GSM follow-on), while 3GPP2 does CDMA
2000
• identified by releases (1999, R4, R5)

May 2001
hgs/SIP Tutorial 132

3G and VoIP

• GPRS not suitable for VoIP: low bandwidth, high delay (500-600 ms RTT)
• initially (R4), CS voice to base station, then ATM/IP packets
• later (R5), in Internet multimedia (IM) subsystem ➠ IP to UE (user equipment)
• uses AMR audio codec, with variable rate of 4.75 to 12.2 kb/s, or GSM HR or EFR
• UTRAN delays: see TR 25.932

May 2001
hgs/SIP Tutorial 133

Signaling in 3GPP IM subsystem

Uses SIP for session setup and defines new entities:

Proxy CSCF: first point of contact in visited network; finds the user’s home network
and provide some translation, security and authorization functions
Serving CSCF: controls sessions, acts as registrar and triggers and executes services.
Accesses the user’s profile; can be located in the home or visited network.
Interrogating CSCF: first point of contact in home network. It assigns the serving
CSCF, contacts the HSS and forwards SIP request.

May 2001
hgs/SIP Tutorial 134

3G SIP registration

mobility management
signaling

serving interrogating interrogating

CSCF

proxy home IM domain

registration signaling (SIP)_

visited IM domain

May 2001
hgs/SIP Tutorial 135

Differences to “standard” SIP

• requires REGISTER before making call

• INVITE uses authentication information provided by REGISTER ➠ Path header
• always visit I/P/S for “home” services
• compression on link from UE to P-CSCF

May 2001
hgs/SIP Tutorial 136

RFCs

draft-ietf-sip-rfc2543bis-03 base protocol spec

RFC 3087 Control of Service Context using SIP Request-URI
RFC 3050 Common Gateway Interface for SIP
RFC 2916 E.164 number and DNS
RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and
Telephony Signals
RFC 2806 URLs for Telephone Calls
RFC 2543 SIP: Session Initiation Protocol

May 2001
hgs/SIP Tutorial 137

For more information. . .

SIP: http://www.cs.columbia.edu/sip
SDP: http://www.cs.columbia.edu/˜hgs/internet/sdp.html
RTP: http://www.cs.columbia.edu/˜hgs/rtp
Papers: http://www.cs.columbia.edu/IRT

May 2001