hgs/SIP Tutorial

The Session Initiation Protocol (SIP)

alireza navabi

May 2001

hgs/SIP Tutorial

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

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

VoIP protocol architecture

Languages/APIs JAIN CPL voiceXML servlets Parlay sipcgi Directory/Discovery DNS/enum LDAP TRIP SLP Signaling SIP SDP MGCP H.248 PINT
SPIRITS

peertopeer

RTSP

masterslave

QoS DiffServ IntServ

Transport RTP TLS SCTP

May 2001

hgs/SIP Tutorial

Multimedia protocol stack

signaling quality of service media transport

MGCP/Megaco

SDP H.323 SIP RTSP

reservation

measurement

RSVP

RTCP

RTP
transport

TCP IPv4, IPv6 PPP AAL3/4 ATM AAL5

UDP

network

PPP Ethernet V.34

link

Sonet
physical

May 2001

kernel

application daemon

media encaps. (H.261, MPEG)

hgs/SIP Tutorial

SIP protocol use

LDAP

address lookup

TRIP PSTN gateway lookup

DNS

nexthop

signaling

may trigger

SIP SDP
sets up

SIP SDP RSVP

RSVP
media

RTP UDP

RTP UDP

May 2001

hgs/SIP Tutorial

SIP applications
setting up voice-over-IP calls setting up multimedia conferences event notication (subscribe/notify) IM and presence text and general messaging signaling transport

May 2001

hgs/SIP Tutorial

SIP addressing

May 2001

hgs/SIP Tutorial

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] [email protected] columbia.edu

tel:12015551234

[email protected]

May 2001

hgs/SIP Tutorial

10

SIP addressing
typically, same as users 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 SIP redirect server SIP stateless proxy SIP (forking) proxy
A@ B@ C@

IP phone, PC, conference bridge returns new location for requests routes call requests routes call requests maintains mappings from names to addresses

SIP registrar

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)

INVITE b2b 200 OK

SIP UA2 (UAC)

INVITE callee

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 yes no no stateless no yes no stateful unlikely common yes call state common possible (rewall) N/A

UA (UAC, UAS) proxy redirect registrar

May 2001

hgs/SIP Tutorial

16

SIP architecture: peer-to-peer

SIP redirect server

user agent (UA)

user agent (UA)

user agent (UA)

Internet
RTP audio

128.119.40.186

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 proxy

Internet

registrar 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?, TRIPGW? sip:[email protected]

sip:12125551234@domain

SIP H.248

tel:+12125551234
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]

IP 010

INVITE sip:[email protected]

May 2001

hgs/SIP Tutorial

20

SIP operation in proxy mode

cs.columbia.edu

? location server
1 2

hgs@play

henning

cs.tuberlin.de

INVITE [email protected] 200 OK

INVITE hgs@play 5
3

200 OK
6

[email protected]

play

tune
8

ACK hgs@play

media stream

May 2001

hgs/SIP Tutorial

21

SIP operation in redirect mode

ieee.org
1
henning

?
2 columbia.edu

location server

tu-berlin.de INVITE [email protected] 302 Moved temporarily Contact: [email protected]

4 5 ACK [email protected]

6 INVITE [email protected] 7 200 OK 8 ACK [email protected]

columbia.edu

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] Contact: alice@pc17 A@ B@ C@ SQL, LDAP, Corba, proprietary, ...

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 ll in details later route using request URIs all but rst 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 rewalls 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 yahoo.com yahoo.com yahoo.com MX MX MX MX 1 1 1 9 mx2.mail.yahoo.com mx3.mail.yahoo.com mx1.mail.yahoo.com 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 _sip._udp SRV SRV SRV SRV 0 1 0 1 0 0 0 0 5060 5060 5060 5060 sip-server.cs.columbia.edu. backup.ip-provider.net. sip-server.cs.columbia.edu. 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 SRV 0 0 s2.example.com SRV 0 0 s3.example.com

b1.example.com, b2.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 cant 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 rst 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 method URL SIP/2.0 response SIP/2.0 status reason

Via: SIP/2.0/ protocol host:port From: user <sip:from_user@source> To: user <sip:to_user@destination> CallID: localid@host CSeq: seq# method ContentLength: length of body ContentType: media type of body Header: parameter ;par1=value ;par2="value" ;par3="value folded into next line" blank line
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

message body

message header

May 2001

hgs/SIP Tutorial

33

SIP syntax
eld names and some tokens (e.g., media type) are case-insensitive everything else is case-sensitive white space doesnt matter except in rst line lines can be folded multi-valued header elds can be combined as a comma-list

May 2001

hgs/SIP Tutorial

34

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

May 2001

hgs/SIP Tutorial

35

SIP invitation and media negotiation

[email protected]
INVITE sip:[email protected] SIP/2.0 From: sip:[email protected] To: sip:[email protected] CallID: [email protected] CSeq: 42 INVITE ContentType: application/sdp v=0 o=user1 536 2337 IN IP4 h3.wonderland.com c=IN IP4 h3.wonderland.com m=audio 3456 RTP/AVP 0 1 m=video 4000 RTP/AVP 38 39

calls

[email protected]
SIP/2.0 200 OK From: sip:[email protected] To: sip:[email protected] CallID: [email protected] CSeq: 42 INVITE ContentType: application/sdp v=0 o=user1 535 687637 IN IP4 m.macrosoft.com c=IN IP4 m.macrosoft.com m=audio 1200 RTP/AVP 1 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
100 Trying 180 Ringing 181 Call forwarded 182 Queued 183 Session Progress

Success
200 OK

Redirection
300 Multiple Choices 301 Moved Perm. 302 Moved Temp. 380 Alternative Serv.

Request Failure
400 Bad Request 401 Unauthorized 403 Forbidden 404 Not Found 405 Bad Method 415 Unsupp. Content 420 Bad Extensions 486 Busy Here

500 Server Error 501 Not Implemented 503 Unavailable 504 Timeout

600 Busy Everwhere 603 Decline 604 Doesnt Exist 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]
Via: y1.yahoo.com Via: a.example.com Via: a.example.com

[email protected]

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 To: sip:[email protected] 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

acme.com

sales.acme.com

[email protected]

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

branch= h(To,From,CallID,CSeq,URL)

INVITE sip:[email protected] SIP/2.0 h(info,alice,17,1,bob@sales) Via: sales.acme.com;branch= Via: acme.com;branch= h(info,alice,17,1,bob@sales) 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: rewalls anonymizer proxies proxies controlling PSTN gateways use Record-Route and Route

May 2001

hgs/SIP Tutorial

43

Request routing
A INVITE PB
Contact: A

PA INVITE PB
Contact: A RecordRoute: PA

PB INVITE B
Contact: A RecordRoute: PB, PA

200 OK
Contact: B RecordRoute: PB, PA

200 OK
Contact: B RecordRoute: PB, PA

200 OK
Contact: B RecordRoute: PB, PA

ACK PA
Route: PB,B

ACK PB
Route: B

ACK B

BYE A

BYE A
Route: A;maddr=PB

BYE A
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 [email protected]

INVITE bob@b

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

q=0.7

INVITE

486 Busy here

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 nal answer = lowest status code forward provisional responses

May 2001

hgs/SIP Tutorial

47

Parallel forking call ow

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 ows 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 BYE 500 ms UAS, proxy

used for BYE, OPTIONS, SUBSCRIBE, NOTIFY, . . . 1xx sent by UAS or proxy only if no nal answer expected within 200 ms if provisional response, retransmit with T 2 (4) seconds

1s

2s 4s 4s ... no more than 11 packets

200, 4xx, 5xx, 6xx

May 2001

hgs/SIP Tutorial

50

INVITE reliability
INVITE is special long time between request and nal response 100 (by proxy) indicates request has been received proxy usually forwards 1xx from all branches only retransmit until 100 ACK conrms receipt of nal response
status ACK event request sent T1*2 INVITE
n

Initial
INVITE

Calling
1xx

status ACK

Call proceeding
status ACK

1xx 7 INVITE sent

Completed

May 2001

hgs/SIP Tutorial

51

Other signaling approaches

May 2001

hgs/SIP Tutorial

52

Differences to classical signaling

name in-band out-of-band IP examples E&M, DTMF ISUP, Q.931 SIP network same different typically same channel same different 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 whats happening can charge for caller id, call waiting

May 2001

hgs/SIP Tutorial

54

MGCP/SIP architecture
STP call agent MG controller
SIP H.323

call agent MG controller

SIP H.323

TCAP

SS7 gwy
ISUP

SCP

MGCP/Megaco

MGCP/Megaco

SS7

RGW TGW RTP RGW

Internet

PSTN

May 2001

hgs/SIP Tutorial

55

Extending SIP
extension new headers new headers new method new body type new status code new URL type behavior ignored mandatory determine? Supported OPTIONS Accept ?

class-based

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 SIP/2.0 420 Bad Extension Unsupported: com.example.billing SIP/2.0 421 Extension Required Require: 183

S->C:

S->C:

May 2001

hgs/SIP Tutorial

57

User identication

May 2001

hgs/SIP Tutorial

58

Standard call/caller identication

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 veried 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 unicast multicast media unicast multicast telephony multicast session reach rst 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

BYE A

REFER B2 ReferredBy: B1

INVITE B2 ReferredBy: B1

B2

May 2001

hgs/SIP Tutorial

65

IVR and VoiceXML

A@ B@ C@

SQL, LDAP

SIP REFER 200

SIP UA VoiceXML VoiceXML scripts text RTP

text

May 2001

hgs/SIP Tutorial

66

Third-party call control

RTP

200 200 2 5 INVITE SDP (from 2) ACK SDP (from 4) INVITE no SDP 4 3 6

SIP SIP

ACK

May 2001

hgs/SIP Tutorial

67

SIP billing/charging
What for? transport resource reservation protocol SIP services (call processing) authentication PSTN gateway services media server services (translation, storage) How? resource reservation protocols SIP-in-DIAMETER approach server log les

May 2001

hgs/SIP Tutorial

68

Security issues

May 2001

hgs/SIP Tutorial

69

Threats
spoong From in REGISTER: call redirection spoong From in INVITE: bypass call ltering snooping media packets billing confusion (identier munging) denial-of-service attacks

May 2001

hgs/SIP Tutorial

70

SIP security
layer/mechanism network layer transport layer SIP INVITE SIP REGISTER SIP general approach IPsec TLS basic/digest basic/digest S/MIME characteristics adjacent nodes, all or nothing, hard to congure adjacent nodes, all or nothing shared secrets with random parties securing headers? 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 Certicate: 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 cant 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 rst, then phone, then answering service) caller proposes, callee disposes

May 2001

hgs/SIP Tutorial

80

Extended SIP Contact header

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

macrosoft.com presence server PA alice@ example.com SUBSCRIBE UA NOTIFY registrar

PUA

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 alice presentity, PA 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 notication rates vary gradual transition to continuous media

IR detector 0.01

temperature sensor 0.1

process control

packet audio/video 100

1 10 events (packets) / second

Event describes event type

May 2001

hgs/SIP Tutorial

91

Example home architecture

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

(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 same as scripting same as Java very same as Java language? any Java XML Java party? callee callee both caller

SIP-cgi servlets CPL applets

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 ofce and lab in parallel, if that fails, try home allow call to mobile if Ive 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 Administrative screening (rewall) Central phone server Intelligent user location

Third-party registration control Calendarbook access Client billing allocation (lawyers ofce) End system busy Phone bank tion/queueing) (call distribu-

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, nal 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 url: sip:jones@ example.com proxy timeout: 10s timeout failure

Call

String-switch field: from match: *@example.com otherwise

location url: sip:jones@ voicemail. example.com merge: clear

redirect

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 subelds 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 specied 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 rst 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 specied location clear: clear earlier location values Only one output; cannot fail. Dont 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 specied in param location: caller preference location to apply

May 2001

hgs/SIP Tutorial

119

Signalling Actions: proxy

Proxy the call to the currently-specied set of locations, and automatically select one best nal 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, rst-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-specied set of locations. This has no specic 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 denes a tree; we want CPLs to be represented as directed acyclic graphs. Subactions are dened at the top level of the script, outside other actions. for acyclicity, top-level actions and subactions may only call subactions which were dened earlier in the script. Anywhere a node is expected, you can instead have a sub tag, with a ref parameter which refers to a subactions 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 dont 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 identied 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 denes new entities: Proxy CSCF: rst point of contact in visited network; nds the users home network and provide some translation, security and authorization functions Serving CSCF: controls sessions, acts as registrar and triggers and executes services. Accesses the users prole; can be located in the home or visited network. Interrogating CSCF: rst 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 CSCF proxy

interrogating

interrogating

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