spring 2004 introduction to sip school of electronics and information kyung hee university choong...
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Spring 2004
Introduction to SIPIntroduction to SIP
School of Electronics and InformationKyung Hee University
Choong Seon [email protected]
http://networking.khu.ac.kr
2Spring 2004
Overview of SIP
Proposed standard released in 1999, current release in 2002
Session-layer control-plane signaling protocol with support for establishing, modifying and terminating one-to-one or multiparty sessions.
Light-weight, ASCII based generic signaling protocol to facilitate multimedia communications over IP
Independent of media characteristics and transport protocol properties
Work in IETF currently by SIP WG and SIPPING WG Related work by MMUSIC WG, AAA WG, GEOPRIV
WG, SIMPLE WG, Internet Telephony WG 3GPP has decided to support it in IP multimedia
services (IMS) domain.
3Spring 2004
Overview of SIP (cont’d)
SIP is complimentary to SGCP/MGCP SIP Provides Session Control SGCP/MGCP Provides Device Control
4Spring 2004
The Big FAQ and SIP?The Big FAQ and SIP?
Q: You are too IP-centric, aren’t you?A: Of course, we are.
Internet telephony (which has Internet in its name) is about IP. IP telephony runs on top of IP and utilizes the IP
service model. It is not about re-engineering PSTN -- PSTN is
good enough. SIP is much more similar to HTTP rather
than to legacy signaling both in terms of service model and protocol design.
5Spring 2004
Problem description
Environment: aging PSTN equipment, carrier fiber overcapacity and arrival of multimedia communications with preparation to millions of mobile hosts that are IP-aware. Additionally, the possibility of programming PSTN-services in a more open environment.
Problem: how to enable gradual deployment and transition to avoid the one-big-leap-for-mankind scenarios?
Approach: IETF multimedia architecture and interoperability schemes with PSTN and 3GPP. Signaling is one part of this. The multimedia architectures capabilities are related to Quality of Service architecture and IP multicast.
6Spring 2004
IETF Multimedia Architecture
7Spring 2004
What Protocols Are Needed?
Signaling protocol to establish presence, locate users, set up, modify and teardown sessions
Media Transport Protocols for transmission of packetized audio/video
Supporting Protocols Gateway Location, QoS, interdomain AAA,
address translation, IP, etc.
AAA = Authentication, Authorization, Accounting
8Spring 2004
What Protocols Are There
Signaling: SIP/SDP (IETF), H.323 (ITU-T) Note: SIP adopted by 3gpp; lower production and operation
costs reported Media: RTP (IETF’s, adopted by ITU-T) Transport: UDP, TCP, (Stream Control Transmission
Protocol – RFC 2960) Supporting protocols:
DNS TRIP - Telephony Routing over IP - discovery and exchang
e of IP telephony gateway routing tables between providers RSVP - Resource Reservation Setup Protocol COPS - Common Open Policy Service - protocol for for sup
porting policy control over QoS Diameter - Authentication, Accounting, Authorization
9Spring 2004
SIP SignalingSIP Signaling
SIP is end-to-end, client-server session protocol SIP’s primarily provides presence and mobility Protocol primitives: Session setup, termination, ch
anges Arbitrary services built on top of SIP, e.g.:
Redirect calls from unknown callers to secretary Reply with a webpage if unavailable Send a JPEG on invitation
Features: Textual encoding (telnet, tcpdump compatible) Programmability
10Spring 2004
SIP - General Purpose Presence Protocol
SIP is not limited to Internet SIP establishes user presence SIP messages can convey arbitrary signaling payload:
session description, instant messages, JPEGs, any MIME types
Suitable for applications having a notion of session distributed virtual reality systems, network games (Quake II/III implementations), video conferencing, etc.
Applications may leverage SIP infrastructure (Call Processing, User Location, Authentication) Instant Messaging and Presence SIP for Appliances
11Spring 2004
Internet MultimediaInternet Multimedia
Real Time Protocol (RTP) – media packets Real Time Control Protocol (RTCP) – monitor & report Session Announcement Protocol (SAP) Session Description Protocol (SDP) Session Initiation Protocol (SIP) Real Time Stream Protocol (RTSP) – play out control Synchronized Multimedia Integration Language (SMIL) – mixes audi
o/video with text and graphics : August 2001
References: Search keyword at http://www.rfc-editor.org/rfc.html
For SMIL - http://www.w3.org/AudioVideo/
12Spring 2004
Stages of IP Signaling Development
Precommercial stage (1980-1995) Research in organizations and universities IETF: Audio/Video Transport (AVT) WG - RTP IETF: Multiparty Multimedia Session Control (MMUSIC) WG – SIP
PC-centric stage (1995-1998) First commercial VoIP software, proprietary software Calls from multimedia PC to another multimedia PC ITU: May-June 1995: H.323v1 Most commercial applications H.323 compliant by the end of 1996
Carrier-grade stage (1998-) Service provider VoIP deployment First obstacle: integration to PSTN signalling (gateway functionality) Development of media gateway controller protocols (MGCP and Me
gaco/H.248) Todays approach: coexistence of SIP, H.323, MGCP and Megaco/
H.248
13Spring 2004
SIP HistorySIP History
Work began in 1995 in IETF mmusic WG 02/1996: draft-ietf-mmusic-sip-00: 15 ASCII pages, one request
type 12/1996: -01 30 ASCII pages, 2 request types 01/1999: -12 149 ASCII pages, 6 methods 03/1999: RFC 2543, 153 ASCII pages, 6 methods 11/1999: SIP WG formed 11/2000: draft-ietf-sip-rfc2543bis-02, 171 ASCII pages, 6 metho
ds 12/2000: it was recognized that amount of work at SIP WG was
becoming unmanageable; 1 RFC; 18 I-Ds on WG’s agenda; numerous individual submissions
04/2001: proposal for splitting SIP WG into SIP and SIPPING announced
2001: SIP implementations widely available http://www.cs.columbia.edu/~hgs/sip/implementations.html http://www.pulver.com/sip/products.html
14Spring 2004
SIP End-devices
User Agent (user application) UA Client (originates calls) UA Server (listens for incoming calls) both SW and HW available
15Spring 2004
SIP ComponentsSIP Components
SIP Proxy Server relays call signaling, i.e. acts as both client and
server operates in a transactional manner, i.e., it keeps
no session state
SIP Redirect Server redirects callers to other servers
SIP Registrar accept registration requests from users maintains user’s whereabouts at a Location
Server (like GSM HLR)
16Spring 2004
Telephony on the InternetTelephony on the Internet may not be a stand-alone business, but part of IP servicesmay not be a stand-alone business, but part of IP services
Public IP Backbone• Goes everywhere• End-to-end control• Consistent for all services
• DNS – mobility• Messaging • Web• Directory• Security• QoS• Media services• Sessions• Telephony• …………
SIP
RTP
CAS, Q.931, SS7
PCM
Telephone Gateway SIP client
MG
SG
MGCP
SIP/RTP Media SIP/RTP Media ArchitectureArchitecture
Any other sessionsAny other sessionsCAS : channel associated signaling
17Spring 2004
Commercial Grade IP TelephonyCommercial Grade IP Telephony
New services (new revenue)
Scalability (Web-like)
Baseline PSTN&PBX features
Client & user authentication
Accounting assured QoS
QoS assured signaling
Security assured signaling
Hiding of caller ID & location
Better than PSTN features• New & fast service creation• Internet (rapid) scalability• Mobility• Dynamic user preferences• End-to-end control
• Service selection• Feature control
• Mid-call control features
• Pre-call• Mid-call
Assure baseline PSTN features
Leverage and Commonality of telephony with the Web/Internet
18Spring 2004
IP CommunicationsIP Communications
PSTN/PBX-like: POTS AIN CS-1, CS-2 PBX & Centrex
User has control of: All addressable devices Caller and called party
preferences
Better quality than 3.1 kHz
Web-like: Presence Voice and text chat Messaging Voice, data, video Multiparty
Conferencing Education Games
Any qualityMost yet to be invented
Complete integration of all services under full user control
19Spring 2004
Development of SIPDevelopment of SIP
IETF - Internet Engineering Task Force MMUSIC - Multiparty Multimedia Session Control Working Grou
p SIP developed by Handley, Schulzrinne, Schooler, and Rosenb
erg Submitted as Internet-Draft 7/97
Assigned RFC 2543 in 3/99 Internet Multimedia Conferencing Architecture.
Alternative to ITU’s H.323 H.323 used for IP Telephony since 1994 Problems: No new services, addressing, features Concerns: scalability, extensibility
20Spring 2004
SIP PhilosophySIP Philosophy
Internet Standard IETF - http://www.ietf.org
Reuse Internet addressing (URLs, DNS, proxies) Utilizes rich Internet feature set
Reuse HTTP coding Text based
Makes no assumptions about underlying protocol: TCP, UDP, X.25, frame, ATM, etc. Support of multicast
21Spring 2004
SIP Clients and Servers - 1SIP Clients and Servers - 1
SIP uses client/server architecture Elements:
SIP User Agents (SIP Phones) SIP Servers (Proxy or Redirect - used to locate SIP users or t
o forward messages.)• Can be stateless or stateful
SIP Gateways:• To PSTN for telephony interworking• To H.323 for IP Telephony interworking
Client - originates message Server - responds to or forwards message
22Spring 2004
SIP Clients and Servers - 2SIP Clients and Servers - 2
Logical SIP entities are: User Agents
User Agent Client (UAC): Initiates SIP requests User Agent Server (UAS): Returns SIP responses
Network Servers Registrar: Accepts REGISTER requests from clients Proxy: Decides next hop and forwards request Redirect: Sends address of next hop back to client
The different network server types may be collocated
23Spring 2004
SIP AddressingSIP Addressing
SIP gives you a globally reachable address Callees bind to this address using SIP REGISTER m
ethod. Callers use this address to establish real-time commu
nication with callees.
URLs used as address data format; examples: sip:[email protected] sip:[email protected]?subject=callme sip:[email protected]; geo.position:=48.54_-123.84_120
24Spring 2004
SIP Addressing (cont’d)SIP Addressing (cont’d)
must include host, may include user name, port number, parameters (e.g., transport), etc.
may be embedded in Webpages, email signatures, printed on your business card, etc.
address space unlimited non-SIP URLs can be used as well (mailto:,
http:, ...)
25Spring 2004
SIP RegistrationSIP Registration
26Spring 2004
SIP Session Setup ExampleSIP Session Setup Example
200 OK
ACK
INVITE sip:[email protected]
host.wcom.com sip.uunet.com
SIPUser Agent
Client
SIPUser Agent
Server
BYE
200 OK
Media Stream
27Spring 2004
Proxy Server ExampleProxy Server Example
server.wcom.com
200 OK
BYE
200 OK
INVITE sip:[email protected]
host.wcom.com
200 OK
ACK
INVITE sip:[email protected]
sip.uunet.com
SIPUser Agent
Client
SIPProxyServer
SIPUser
AgentServer
Media Stream
28Spring 2004
SIP Operation in Proxy ModeSIP Operation in Proxy Mode
29Spring 2004
Proxy Server Functionality
Serve as rendezvous point at which callees are glabally reachable
Perform routing function, i.e., determine to which hop (UA/proxy/redirect) signaling should be relayed
Allow the routing function to be programmable. Arbitrary logic may be built on top of the protocol user’s signaling preferences AAA firewall control etc.
Forking: Several destinations may be tried for a request sequentially or in parallel.
30Spring 2004
Proxy ChainingProxy Chaining
There may be also cases when a local outbound proxy may be involved provides locally important call processing logic (e.g., identify
ing nearest 119) manages firewall provides least-gateway-cost routing service IP phones must know address of the proxy:may be configur
ed manually or with a configuration protocol (DHCP, TFTP, ... )
In general, servers may be arbitrarily chained a central company’s server may distribute signaling to depa
rtmental servers a user may want to forward incoming calls to her cell phone
31Spring 2004
Proxy Chaining – an ExampleProxy Chaining – an Example
Note : signaling (in red) may take a completely different path from media in blue)
32Spring 2004
Redirect Server Example
302 Moved sip:[email protected]
ACK
Media Stream
INVITE sip:[email protected]
SIPUser Agent
Client
SIPRedirectServer
180 Ringing
ACK
INVITE sip:[email protected]
SIPUser Agent
ServerREGISTER [email protected]
host.wcom.com sip.uunet.com
200 OK
server.wcom.com
200 OK
CC
RS
UAS
1
2
3
33Spring 2004
SIP Operation in Redirect Mode
34Spring 2004
SIP Server - Proxy versus Redirection
A SIP server may either proxy or redirect a request Which of the two method applies is a configuration i
ssue. It may be statically configured or dynamically determined/
Redirection useful if a user moves or changes her provider (PSTN: “The number you have dialed is not available.”) -- caller does not need to try the original server next time. Stateless.
Proxy useful if forking, AAA, firewall control needed. In general, proxying grants more control to the server.
35Spring 2004
SIP Requests : RFC2543 MethodsSIP Requests : RFC2543 Methods
SIP Requests (Messages) defined as: Method SP Request-URI SP SIP-Version CRLF (SP=Space, CRLF=Carriage Retu
rn and Line Feed)
Example: INVITE sip:[email protected] SIP/2.0
Method Description
INVITE A session is being requested to be setup using a specified media
ACK Message from client to indicate that a successful response to an INVITE has been received
OPTIONS A Query to a server about its capabilities
BYE A call is being released by either party
CANCEL Cancels any pending requests. Usually sent to a Proxy Server to cancel searches
REGISTER Used by client to register a particular address with the SIP server; binds a permanent address to current location
36Spring 2004
SIP Requests ExampleSIP Requests Example
Required Headers (fields):
Via: Shows route taken by request. Call-ID: unique identifier generated by client. CSeq: Command Sequence number
• generated by client• Incremented for each successive request
INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP host.wcom.com:5060From: Alan Johnston <sip:[email protected]>To: Jean Luc Picard <sip:[email protected]>Call-ID: [email protected]: 1 INVITE
}Uniquely identify this session request
37Spring 2004
SIP Requests ExampleSIP Requests Example
Typical SIP Request:
INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP host.wcom.com:5060From: Alan Johnston <sip:[email protected]>To: Jean Luc Picard <sip:[email protected]>Call-ID: [email protected]: 1 INVITEContact: sip:[email protected]: Where are you these days?Content-Type: application/sdp Content-Length: 124
v=0o=ajohnston 5462346 332134 IN IP4 host.wcom.coms=Let's Talkt=0 0c=IN IP4 10.64.1.1m=audio 49170 RTP/AVP 0 3
38Spring 2004
SIP ResponsesSIP Responses
SIP Responses defined as (HTTP-style): SIP-Version SP Status-Code SP Reason-Phrase CRLF (SP=Space,
CRLF=Carriage Return and Line Feed)
Example: SIP/2.0 404 Not Found First digit gives Class of response:
Description Examples
1xx Informational – Request received, continuing to process request.
180 Ringing 181 Call is Being Forwarded
2xx Success – Action was successfully received, understood and accepted.
200 OK
3xx Redirection – Further action needs to be taken in order to complete the request.
300 Multiple Choices 302 Moved Temporarily
4xx Client Error – Request contains bad syntax or cannot be fulfilled at this server.
401 Unauthorized 408 Request Timeout
5xx Server Error – Server failed to fulfill an apparently valid request.
503 Service Unavailable 505 Version Not Suported
6xx Global Failure – Request is invalid at any server. 600 Busy Everywhere 603 Decline
39Spring 2004
SIP Responses Example SIP Responses Example
Required Headers:
Via, From, To, Call-ID, and CSeq are copied exactly from Request.
To and From are NOT swapped!
SIP/2.0 200 OKVia: SIP/2.0/UDP host.wcom.com:5060From: Alan Johnston <sip:[email protected]>To: Jean Luc Picard <sip:[email protected]>Call-ID: [email protected]: 1 INVITE
40Spring 2004
SIP Responses Example SIP Responses Example
Typical SIP Response (containing SDP)
SIP/2.0 200 OKVia: SIP/2.0/UDP host.wcom.comFrom: Alan Johnston <sip:[email protected]>To: Jean Luc Picard <sip:[email protected]>Call-ID: [email protected]: 1 INVITEContact: sip:[email protected]: Where are you these days?Content-Type: application/sdp Content-Length: 107
v=0o=picard 124333 67895 IN IP4 uunet.coms=Engage!t=0 0c=IN IP4 11.234.2.1m=audio 3456 RTP/AVP 0
41Spring 2004
C
Forking Proxy ExampleForking Proxy Example
sip.mci.com
ACK
INVITE
INVITE
404 Not Found
180 Ringing
INVITE sip:[email protected]
host.wcom.com
180 Ringing
ACK
sip.uunet.com
SIPUser Agent
Client
SIPProxyServer
SIPUser AgentServer 2
SIPUser AgentServer 1
proxy.wcom.com
100 Trying
BYE
200 OK
Media Stream
200 OK200 OK
S1
S2
ForkFork
42Spring 2004
SIP Headers - Partial ListSIP Headers - Partial List
Header Description Examples
Accept Indicates acceptable formats. Accept: application/ sdp Accept: currency/ dollars
Authorization Contains encryption information Authorization: pgp info…
Call-ID Used to uniquely identify a particular session or registration messages. Should have randomness to ensure overall global uniqueness.
Call-ID: [email protected] Call-ID: Jan-01-1999-1510- [email protected] i: [email protected]
Contact Alternative SIP URL for more direct message routing.
Contact: W. Riker, Acting Captain <[email protected]> Contact: [email protected]; expires=3600 m: [email protected]
Content-Length Octet count in message body. Content-Length: 285
Content-Type Content type of message body Content-Type: application/ sdp c: application/ h.323
CSeq Command Sequence number – used to distinguish different requests during the same session.
CSeq: 1 INVITE CSeq: 1000 INVITE CSeq: 4325 BYE CSeq: 1 REGISTER
Encryption Encryption information. Encryption: pgp info…
Expires Used to indicate when the message content is no longer valid. Can be a number of seconds or a date and time.
Expires: 60 Expires: Thu, 07 Jan 1999 17:00 CST
43Spring 2004
SIP Headers - Continued SIP Headers - Continued
From Required field containing the originating SIP URL. Can also include a display name.
From: Dana Scully <sip:[email protected]> From: sip:[email protected]; tag=1234567 f: sip: [email protected]
Max-Forwards Count decremented by each server forwarding the message. When goes to zero, server sends a 483 Too Many Hops response.
Max-Forwards: 10
Priority Can specify message priority Priority: normal Priority: emergency
Record-Route Added to a request by a proxy that needs to be in the path of future messages.
Record Route: sip.mci.com
Require Indicates options necessary for the session. Require: local.telephony
Response-Key Contains PGP key for encrypted response expected.
Response-Key: pgp info…
Retry-After Indicates when the resource may be available. Can be a number of seconds or a date and time.
Retry-After: 3600 Retry-After: Sat, 01 Jan 2000 00:01 GMT
44Spring 2004
SIP Headers - ContinuedSIP Headers - Continued
Route Determines the route taken by a message.
Route: orinoco.brooks.net
Subject Can be used to indicate nature of call.
Subject: More about SIP s: You’d better answer!
To Required field containing the recipient SIP URL. May contain a display name.
To: Fox Mulder <sip:[email protected]> To: sip:[email protected]; tag=314 t: sip:[email protected]; tag=52
Unsupported Lists features not supported by server.
Unsupported: tcap.telephony
Via Used to show the path taken by the request.
Via: SIP/ 2.0/ UDP sip.mfs.com Via: SIP/ 2.0/ TCP uunet.com Via: SIP/ 2.0/ UDP 192.168.1.1
Warning Contains a code and text to warn about a problem
Warning: 331 Unicast not available
45Spring 2004
Via Headers and RoutingVia Headers and Routing
Via headers are used for routing SIP messages Requests
Request initiator puts address in Via header Servers check Via with sender’s address, then add own
address, then forward. (if different, add “received” parameter)
Responses Response initiator copies request Via headers. Servers check Via with own address, then forward to next Via address
46Spring 2004
SIP Firewall ConsiderationsSIP Firewall Considerations
Firewall Problem Can block SIP packets Can change IP addresses of packets
TCP can be used instead of UDP Record-Route can be used:
ensures Firewall proxy stays in path A Firewall proxy adds Record-Route header
Clients and Servers copy Record-Route and put in Route header for all messages
47Spring 2004
SIP Message BodySIP Message Body
Message body can be any protocol Most implementations:
SDP - Session Description Protocol RFC 2327 4/98 by Handley and Jacobson
• http://www.ietf.org/rfc/rfc2327.txt Used to specify info about a multi-media session. SDP fields have a required order For RTP - Real Time Protocol Sessions:
• RTP Audio/Video Profile (RTP/AVP) payload descriptions are often used
48Spring 2004
Session Description Protocol (SDP)Session Description Protocol (SDP)
Convey sufficient information to enable participation in a multimedia session
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
Note: indeed SDP is a data format rather than a protocol.
49Spring 2004
SDP ExamplesSDP Examples
v=0
o=sisalem 28908044538 289080890 IN IP4 193.175.132.118
s=SIP Tutorial
c=IN IP4 126.16.69.4
t=28908044900 28908045000
m=audio 49170 RTP/AVP 0 98
a=rtpmap:98 L16/11025/2
50Spring 2004
SDP Examples (cont’d)SDP Examples (cont’d)
SDP Example
v=0o=ajohnston +1-613-555-1212 IN IP4 host.wcom.coms=Let's Talkt=0 0c=IN IP4 101.64.4.1m=audio 49170 RTP/AVP 0 3
SDP Example
v=0o=picard 124333 67895 IN IP4 uunet.coms=Engage!t=0 0c=IN IP4 101.234.2.1m=audio 3456 RTP/AVP 0
Field Descripton
Version v=0
Origin o=<username> <session id> <version> <network type> <address type> <address>
Session Name s=<session name>
Times t=<start time> <stop time>
Connection Data c=<network type> <address type> <connection address>
Media m=<media> <port> <transport> <media format list>
51Spring 2004
Another SDP Example (cont’d) Another SDP Example (cont’d)
v=0o=alan +1-613-1212 IN host.wcom.coms=SSE University Seminar - SIPi=Audio, Listen onlyu=http://sse.mcit.com/university/[email protected]=+1-329-342-7360c=IN IP4 10.64.5.246b=CT:128t=2876565 2876599m=audio 3456 RTP/AVP 0 3a=type:recvonly
52Spring 2004
Authentication & EncryptionAuthentication & Encryption
SIP supports a variety of approaches: end to end encryption hop by hop encryption
Proxies can require authentication: Responds to INVITEs with 407 Proxy-Authentication Required
Client re-INVITEs with Proxy-Authorization header. SIP Users can require authentication:
Responds to INVITEs with 401 Unathorized Client re-INVITEs with Authorization header
53Spring 2004
SIP Encryption Example SIP Encryption Example
INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP host.wcom.com:5060From: Alan Johnston <sip:[email protected]>To: Jean Luc Picard <sip:[email protected]>Call-ID: [email protected]: 1 INVITEContent-Length: 224Encryption: PGP version=2.6.2, encoding=ascii
q4aspdoCjh32a1@WoiLuaE6erIgnqD3erDg8aFs8od7idf@ hWjasGdg,ddgg+fdgf_ggEO;ALewAKFeJqAFSeDlkjhasdfkj!aJsdfasdfKlfghgasdfasdfa|Gsdf>a!sdasdf3w29451k45mser?we5y;343.4kfj2ui2S8~&djGO4kP%Hk#(Khujefjnjmbm.sd;da’l;12’;123=]aw;erwAo3529ofgk
54Spring 2004
PSTN Features with SIPPSTN Features with SIP
Features implemented by SIP Phone Call answering: 200 OK sent Busy: 483 Busy Here sent Call rejection: 603 Declined sent Caller-ID: present in From header Hold: a re-INVITE is issued with IP Addr =0.0.0.0 Selective Call Acceptance: using From, Priority, and Subject headers
Camp On: 181 Call Queued responses are monitored until 200 OK is sent by the called party
Call Waiting: Receiving alerts during a call
55Spring 2004
PSTN Features with SIPPSTN Features with SIP
Features implemented by SIP Server Call Forwarding: server issues 301 Moved Permanently or 302 Moved Temporarily response with Contact info
Forward Don’t Answer: server issues 408 Request Timeout response
Voicemail: server 302 Moved Temporarily response with Contact of Voicemail Server
Follow Me Service: Use forking proxy to try multiple locations at the same time
Caller-ID blocking - Privacy: Server encrypts From information
56Spring 2004
SIP User Location ExampleSIP User Location Example
Q=Q=qualityquality gives preference gives preference
SIP/2.0 302 Moved temporarilySIP/2.0 302 Moved temporarily
Contact: sip:[email protected]: sip:[email protected]
;service=IP,voice mail;service=IP,voice mail
;media=audio ;duplex=full ;q=0.7;media=audio ;duplex=full ;q=0.7
Contact : phone: +1-972-555-1212; service=ISDNContact : phone: +1-972-555-1212; service=ISDN
;mobility=fixed; language=en,es, ;q=0.5;mobility=fixed; language=en,es, ;q=0.5
Contact : phone: +1-214-555-1212; service=pagerContact : phone: +1-214-555-1212; service=pager
;mobility=mobile;mobility=mobile
;duplex=send-only ;media=text; q=0.1; priority=urgent;duplex=send-only ;media=text; q=0.1; priority=urgent
;description=“For emergency only”;description=“For emergency only”
Contact : mailto: [email protected] : mailto: [email protected]
SIP supports mobility across networks and devicesSIP supports mobility across networks and devices
57Spring 2004
Programming SIPProgramming SIP
Examples “discard all calls from Monica during my business
hours” “redirect authenticated friends to my cell phone, a
nyone else to my secretary” “if busy, return my homepage and redirect to reco
rder”
Users and third parties may program SIP follows HTTP programming model Mechanisms suggested in IETF: CGI, Call P
rocessing Language (CPL), Servlets
58Spring 2004
Call Processing Logic ExampleCall Processing Logic Example
59Spring 2004
SIP Mobility SupportSIP Mobility Support
SIP RedirectServer
SIP ProxyServerForeign
Network
MobileHost
HomeNetwork
CorrespondingHost
1 23
6
4
5
1 INVITE
2 302 moved temporarily
3, 4 INVITE
5, 6 OK
7 Data
Global: Wire and wireless
No tunneling required
No change to routing
For fast hand-offs use:
• Use Cellular IP or
• Use DRCP
7
* Dynamic Registration and Configuration Protocol
60Spring 2004
SIP MobilitySIP Mobility
Pre-call mobility MH can find SIP server via
multicast REGISTER MH acquires IP address via
DHCP MH updates home SIP
server
Mid-call mobility MH->CH: New INVITE with
Contact and updated SDP Re-registers with home
registrar
Need not bother home registrar: Use multi-stage registration
Recovery from disconnects
61Spring 2004
Mobile IP CommunicationsMobile IP Communications
Mobile IP Requirements Transparency above L2:Move but keep IP address and all se
ssions alive Mobility
Within subnet Within domain Global
AAA and NAIs Location privacy QoS for r.t. communications
Evolution of Wireless Mobility Circuit Switched Mobility
based on central INs
LAN-MAN: Cellular IP Wide Area: Mobile IP Universal (any net): SIP
62Spring 2004
Presence, Instant Messaging and VoicePresence, Instant Messaging and Voice
http://www.ietf.org/internet-drafts/draft-ietf-impp-model-03.txt
63Spring 2004
IP SIP Phones and AdaptorsIP SIP Phones and Adaptors
1
2
3
Are Internet hosts
• Choice of application
• Choice of server
• IP appliance
Implementations
• 3Com (2)
• Cisco
• Columbia University
• Mediatrix (1)
• Nortel (3)
• Pingtel
64Spring 2004
H.323/SIP ComparisonH.323/SIP Comparison
65Spring 2004
H.323 vs. SIP: Basic Call Control
66Spring 2004
H.323 vs. SIP: Advanced features
67Spring 2004
H.323 vs. SIP Scalability
68Spring 2004
H.323 vs. SIP Extensibility of functionality
69Spring 2004
H.323 vs. SIP Ease of customization
70Spring 2004
H.323 vs. SIP Ease of customization
71Spring 2004
H.323 vs. SIP Ease of Implementation
72Spring 2004
Summary :SIP vs H.323Summary :SIP vs H.323
73Spring 2004
Internet DraftsInternet Drafts
Session Timers in the Session Initiation Protocol (SIP) Caller Preferences for the Session Initiation Protocol (SIP) Guidelines for Authors of Extensions to the Session Initiation Protocol (SIP) The Stream Control Transmission Protocol as a Transport for for the Session Initiation Prot
ocol The Session Inititation Protocol (SIP) 'Replaces' Header The SIP Referred-By Mechanism Compressing the Session Initiation Protocol Session Initiation Protocol Extension to Assure Congestion Safety A Mechanism for Content Indirection in Session Initiation Protocol (SIP) Messages The Session Inititation Protocol (SIP) 'Join' Header SIP Authenticated Identity Body (AIB) Format S/MIME AES Requirement for SIP An Extension to the Session Initiation Protocol for Request History Information Communications Resource Priority for the Session Initiation Protocol (SIP)
Indicating User Agent Capabilities in the Session Initiation Protocol (SIP) Connection Reuse in the Session Initiation Protocol (SIP) The Internet Assigned Number Authority Universal Resource Identifier Parameter Registry f
or the Session Initiation Protocol The Internet Assigned Number Authority Header Field Parameter Registry for the Session I
nitiation Protocol Session Initiation Protocol (SIP) Extension for Event State Publication Interactions of Preconditions with Session Mobility in the Session Initiation Protocol (SIP) Obtaining and Using Globally Routable User Agent (UA) URIs (GRUU) in the Session Initiat
ion Protocol (SIP)
74Spring 2004
RFCsRFCs The SIP INFO Method (RFC 2976) MIME media types for ISUP and QSIG Objects (RFC 3204) SIP: Session Initiation Protocol (RFC 3261) Reliability of Provisional Responses in SIP (RFC 3262) SIP: Locating SIP Servers (RFC 3263) SIP-Specific Event Notification (RFC 3265) DHCP Option for SIP Servers (RFC 3361) Hypertext Transfer Protocol (HTTP) Digest Authentication Using Authentication and Key Ag
reement (AKA) (RFC 3310) The Session Initiation Protocol UPDATE Method (RFC 3311) Integration of Resource Management and SIP (RFC 3312) Internet Media Type message/sipfrag (RFC 3420) A Privacy Mechanism for the Session Initiation Protocol (SIP) (RFC 3323) Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trust
ed Networks (RFC 3325) Session Initiation Protocol Extension for Instant Messaging (RFC 3428) The Reason Header Field for the Session Initiation Protocol (SIP) (RFC 3326)
Session Initiation Protocol Extension for Registering Non-Adjacent Contacts (RFC 3327) Security Mechanism Agreement for the Session Initiation Protocol (SIP) Sessions (RFC 33
29) Private Session Initiation Protocol (SIP)Extensions for Media Authorization (RFC 3313)
The Session Initiation Protocol (SIP) Refer Method (RFC 3515) Dynamic Host Configuration Protocol (DHCPv6)Options for Session Initiation Protocol (SIP)
Servers (RFC 3319) An Extension to the Session Initiation Protocol (SIP) for Symmetric Response Routing (RF
C 3581) Session Initiation Protocol Extension Header Field for Service Route Discovery During Regi
stration (RFC 3608)
75Spring 2004
Relevant IETF Working GroupsRelevant IETF Working Groups
Audio/Video Transport (avt) - RTP Differentiated Services (diffserv) – QoS in backbone IP Telephony (iptel) – CPL, GW location, TRIP Integrated Services (intserv) – end-to-end QoS Media Gateway Control (megaco) – IP telephony gateways Multiparty Multimedia Session Control (mmusic) – SIP, SDP, conferencing PSTN and Internet Internetworking (pint) – mixt services Resource Reservation Setup Protocol (rsvp) Service in the PSTN/IN Requesting InTernet Service (spirits) Session Initiation Protocol (sip) – signaling for call setup Signaling Transport (sigtran) – PSTN signaling over IP Telephone Number Mapping (enum) – surprises ! Instant Messaging and Presence Protocol (impp)
http://ietf.org/html.charters/wg-dir.html
76Spring 2004
SIP SummarySIP Summary
SIP is: Relatively easy to implement Gaining vendor and carrier acceptance Very flexible in service creation Extensible and scaleable Appearing in products right now
SIP is not: Going to make PSTN interworking easy Going to solve all IP Telephony issues (QoS)
77Spring 2004
ReferencesReferences
Book on “Internetworking Multimedia” by Jon Crowcroft, Mark Handley, Ian Wakeman, UCL Press, 1999 by Morgan Kaufman (USA) and Taylor Francis (UK)
RFC 3261: “SIP: Session Initiation Protocol”http://www.ietf.org/rfc/rfc3261.txt
The IETF SIP Working Group home page http://www.ietf.org/html.charters/sip-charter.html
SIP Home Page http://www.cs.columbia.edu/~hgs/sip/
Papers on IP Telephony http://www.cs.columbia.edu/sip/papers.html