Internet Telephony

based on SIP

SMU - Dallas
April 28, May 1, 2000
Henry Sinnreich, MCI WorldCom
Alan Johnston, MCI WorldCom
 Internet 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 audio/video with text and graphics

References: Search keyword at http://www.rfc-editor.org/rfc.html

For SMIL - http://www.w3.org/AudioVideo/

2
 Telephony on the Internet
may not be a stand-alone business, but part of IP services

SIP/RTP Media Architecture

Public IP Backbone
Telephone
• Goes everywhere
Gateway • End-to-end control
SIP client • Consistent for all services
CAS, Q.931, SS7 • DNS – mobility
• Messaging
SG SIP
• Web
PCM • Directory
MG RTP
• Security
• QoS
MGCP
• Media services
• Sessions
• Telephony
• …………

Any other sessions 3

 Commercial Grade IP Telephony

Assure baseline PSTN features

Leverage and Commonality of telephony with the Web/Internet

New services (new revenue) Better than PSTN features

Scalability (Web-like)
• New & fast service creation
• Internet (rapid) scalability
Baseline PSTN&PBX features
• Mobility
• Client & user authentication • Dynamic user preferences
• Accounting assured QoS • End-to-end control
• QoS assured signaling • Service selection
• Security assured signaling • Feature control
• Hiding of caller ID & location
• Mid-call control features
• Pre-call
• Mid-call
4
 Internet End-to-End Control

No single point of failure Elective Elective

Server Server User has control of all
All services enabled by applications and choice
protocols: From ftp to web of servers
R Internet R
USER R
R USER

“Dumb Network”

Services supported by ITU Intelligent Network Control:

interfaces and central POTS, ISDN, BISDN, FR, ATM, H.323, MEGACO/H.248, GSM
controllers
User has little
Central Central Central
Control Control Control control

USER SW SW SW SW SW SW USER
UNI NNI NNI UNI
5
 SIP vs. flavors of IPDC, SGSP, MGCP, MEGACO, H.248
(Internet Client-Server vs. Telco Master-Slave Protocols)

CAS, Q.931, SS7 SIP, H.323

GC
PSTN MCGP Internet
PCM RTP
Legend MG
CG: gateway Controller
MG: Media Gateway 1. IP Telephony Gateway
Absorbs PSTN complexity at the edge of IP
GC GC

MCGP RG MCGP
PSTN PSTN
MG MG

IP ?
Internet
TR 303…
2. “Softswitch” a la IN 3. Residential GWY
•
phone to phone only •
breaks e-2-e control model
•
PSTN services •
no services integration
•
single vendor solution •
no choice of server and apps
•
“unequal access” is reinvented 6
 IP Communications

Complete integration of all services under full user control

PSTN/PBX-like: Web-like:
• POTS
• Presence
• Voice and text chat
• AIN CS-1, CS-2
• Messaging
• PBX & Centrex • Voice, data, video
User has control of: • Multiparty
• All addressable devices  Conferencing

 Education
• Caller and called party
 Games
preferences
Any quality
Better quality than 3.1 kHz
Most yet to be invented

Mixt Internet-PSTN: Click’nConnect, ICW, unified messaging

7
 Development of SIP
• IETF - Internet Engineering Task Force
– MMUSIC - Multiparty Multimedia Session Control Working
Group
– SIP developed by Handley, Schulzrinne, Schooler, and
Rosenberg
 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

8
 SIP 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

9
 SIP 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 to 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

10
 SIP 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

11
 SIP Addressing
Uses Internet URLs
– Uniform Resource Locators
– Supports both Internet and PSTN addresses
– General form is name@domain
– To complete a call, needs to be resolved down to User@Host
– Examples:
sip:[email protected]
sip:J.T. Kirk <[email protected]>
sip:[email protected];user=phone
sip:[email protected]
sip:[email protected];phone-context=VNET

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SIP Session Setup Example

SIP SIP
User Agent User Agent
Client Server
INVITE sip:[email protected]

200 OK

ACK

Media Stream

BYE

200 OK

host.wcom.com sip.uunet.com

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 Proxy Server Example

SIP SIP SIP

User Agent Proxy User Agent
Client Server
Server

INVITE sip:[email protected]
INVITE sip:[email protected]

200 OK
200 OK

ACK

Media Stream

BYE

200 OK

host.wcom.com server.wcom.com sip.uunet.com

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 Redirect Server Example
SIP SIP
SIP
User Agent Redirect User Agent
Client Server Server
REGISTER [email protected]

200 OK
INVITE sip:[email protected]

302 Moved sip:[email protected]

1 RS
ACK 2
C
INVITE sip:[email protected] 3
UAS
180 Ringing

200 OK

ACK

Media Stream

host.wcom.com server.wcom.com sip.uunet.com

15
 SIP Requests
SIP Requests (Messages) defined as:
– Method SP Request-URI SP SIP-Version CRLF (SP=Space,
CRLF=Carriage Return 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

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 SIP Requests Example

Required Headers (fields):

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP host.wcom.com:5060

}
From: Alan Johnston <sip:[email protected]> Uniquely
identify
To: Jean Luc Picard <sip:[email protected]> this
Call-ID: [email protected] session
CSeq: 1 INVITE request

–
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
17
 SIP Requests Example
Typical SIP Request:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP host.wcom.com:5060
From: Alan Johnston <sip:[email protected]>
To: Jean Luc Picard <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: sip:[email protected]
Subject: Where are you these days?
Content-Type: application/sdp
Content-Length: 124

v=0
o=ajohnston 5462346 332134 IN IP4 host.wcom.com
s=Let's Talk
t=0 0
c=IN IP4 10.64.1.1
m=audio 49170 RTP/AVP 0 3
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 SIP 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

Informational – Request received, continuing to 180 Ringing

1xx
process request. 181 Call is Being Forwarded

Success – Action was successfully received,

2xx 200 OK
understood and accepted.

Redirection – Further action needs to be taken in 300 Multiple Choices

3xx
order to complete the request. 302 Moved Temporarily

Client Error – Request contains bad syntax or cannot 401 Unauthorized

4xx
be fulfilled at this server. 408 Request Timeout

Server Error – Server failed to fulfill an apparently 503 Service Unavailable

5xx
valid request. 505 Version Not Suported
600 Busy Everywhere
6xx Global Failure – Request is invalid at any server.
603 Decline
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 SIP Responses Example

Required Headers:

SIP/2.0 200 OK
Via: SIP/2.0/UDP host.wcom.com:5060
From: Alan Johnston <sip:[email protected]>
To: Jean Luc Picard <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 INVITE

– Via, From, To, Call-ID, and CSeq

are copied exactly from Request.
– To and From are NOT swapped!
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 SIP Responses Example
Typical SIP Response (containing SDP)
SIP/2.0 200 OK
Via: SIP/2.0/UDP host.wcom.com
From: Alan Johnston <sip:[email protected]>
To: Jean Luc Picard <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: sip:[email protected]
Subject: Where are you these days?
Content-Type: application/sdp
Content-Length: 107

v=0
o=picard 124333 67895 IN IP4 uunet.com
s=Engage!
t=0 0
c=IN IP4 11.234.2.1
m=audio 3456 RTP/AVP 0 21
 Forking Proxy Example
SIP SIP SIP SIP
User Agent Proxy User Agent User Agent
Client INVITE Server Server 1 Server 2
sip:[email protected] INVITE S1
INVITE
C
100 Trying 404 Not Found
ACK S2
Fork
180 Ringing
180 Ringing
200 OK
200 OK

ACK

Media Stream

BYE

200 OK

host.wcom.com proxy.wcom.com sip.mci.com sip.uunet.com

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 SIP 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 Call-ID: [email protected]

particular session or registration Call-ID: Jan-01-1999-1510- [email protected]
messages. Should have i: [email protected]
randomness to ensure overall
global uniqueness.

Contact Alternative SIP URL for more Contact: W. Riker, Acting Captain <[email protected]>
direct message routing. 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 – CSeq: 1 INVITE
used to distinguish different CSeq: 1000 INVITE
requests during the same session. CSeq: 4325 BYE
CSeq: 1 REGISTER

Encryption Encryption information. Encryption: pgp info…

Expires Used to indicate when the Expires: 60

message content is no longer Expires: Thu, 07 Jan 1999 17:00 CST
valid. Can be a number of
seconds or a date and time.

23
 SIP Headers - Continued
From Required field containing the originating SIP From: Dana Scully <sip:[email protected]>
URL. Can also include a display name. From: sip:[email protected];
tag=1234567
f: sip: [email protected]

Max-Forwards Count decremented by each server Max-Forwards: 10

forwarding the message. When goes to
zero, server sends a 483 Too Many Hops
response.
Priority Can specify message priority Priority: normal
Priority: emergency
Record-Route Added to a request by a proxy that needs to Record Route: sip.mci.com
be in the path of future messages.
Require Indicates options necessary for the session. Require: local.telephony

Response-Key Contains PGP key for encrypted response Response-Key: pgp info…
expected.
Retry-After Indicates when the resource may be Retry-After: 3600
available. Can be a number of seconds or a Retry-After: Sat, 01 Jan 2000 00:01 GMT
date and time.

24
 SIP Headers - Continued

Route Determines the route Route: orinoco.brooks.net

taken by a message.

Subject Can be used to indicate Subject: More about SIP

nature of call. s: You’d better answer!

To Required field containing To: Fox Mulder

the recipient SIP URL. May <sip:[email protected]>
contain a display name. To: sip:[email protected];
tag=314
t:
sip:[email protected];
tag=52
Unsupported Lists features not Unsupported: tcap.telephony
supported by server.

Via Used to show the path Via: SIP/2.0/UDP sip.mfs.com

taken by the request. Via: SIP/2.0/TCP uunet.com
v: SIP/2.0/UDP 192.168.1.1
Warning Contains a code and text to Warning: 331 Unicast not available
warn about a problem

25
 Via 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
26
 SIP 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

27
 SIP 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

28
 SDP Examples
SDP Example 1

v=0
o=ajohnston +1-613-555-1212 IN IP4
host.wcom.com
s=Let's Talk
Field Descripton
t=0 0
c=IN IP4 101.64.4.1 Version v=0
m=audio 49170 RTP/AVP 0 3 o=<username> <session id> <version>
Origin
<network type> <address type> <address>

SDP Example 2 Session Name s=<session name>

Times t=<start time> <stop time>

v=0 c=<network type> <address type>
Connection Data
o=picard 124333 67895 IN IP4 <connection address>
uunet.com m=<media> <port> <transport> <media
s=Engage! Media
format list>
t=0 0
c=IN IP4 101.234.2.1
m=audio 3456 RTP/AVP 0
29
Another SDP Example

v=0
o=alan +1-613-1212 IN host.wcom.com
s=SSE University Seminar - SIP
i=Audio, Listen only
u=http://sse.mcit.com/university/
[email protected]
p=+1-329-342-7360
c=IN IP4 10.64.5.246
b=CT:128
t=2876565 2876599
m=audio 3456 RTP/AVP 0 3
a=type:recvonly

30
 Authentication & 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
31
 SIP Encryption Example

INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP host.wcom.com:5060
From: Alan Johnston <sip:[email protected]>
To: Jean Luc Picard <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 224
Encryption: PGP version=2.6.2, encoding=ascii

q4aspdoCjh32a1@WoiLuaE6erIgnqD3erDg8aFs8od7idf@
hWjasGdg,ddgg+fdgf_ggEO;ALewAKFeJqAFSeDlkjhasdf
kj!aJsdfasdfKlfghgasdfasdfa|Gsdf>a!
sdasdf3w29451k45mser?
we5y;343.4kfj2ui2S8~&djGO4kP
%Hk#(Khujefjnjmbm.sd;da’l;12’;123=]aw;erwAo3529
ofgk
32
 PSTN 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

33
 PSTN 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
34
 SIP User Location Example
SIP supports mobility across networks and devices
Q=quality gives preference

SIP/2.0 302 Moved temporarily

Contact: sip:[email protected]
;service=IP,voice mail
;media=audio ;duplex=full ;q=0.7
Contact : phone: +1-972-555-1212; service=ISDN
;mobility=fixed; language=en,es, ;q=0.5
Contact : phone: +1-214-555-1212; service=pager
;mobility=mobile
;duplex=send-only ;media=text; q=0.1; priority=urgent
;description=“For emergency only”
Contact : mailto: [email protected]
35
 SIP Mobility Support

4
Mobile 5 SIP Redirect
Host Server

SIP Proxy
Foreign Server Home
Network Network
3
7 1 2 6
1 INVITE
Corresponding Global: Wire and wireless
2 302 moved temporarily
Host No tunneling required
3, 4 INVITE No change to routing
5, 6 OK For fast hand-offs use:
7 Data • Use Cellular IP or
• Use DRCP
36
 SIP Mobility

Pre-call mobility Mid-call mobility

• MH can find SIP server • MH->CH: New INVITE
via multicast REGISTER with Contact and
• MH acquires IP address updated SDP
via DHCP • Re-registers with home
• MH updates home SIP registrar
server

Need not bother home registrar: Use multi-stage registration

Recovery from disconnects

37
 Mobile IP Communications

Mobile IP Requirements Evolution of Wireless Mobility

• Transparency above L2: • Circuit Switched Mobility
Move but keep IP address and
all sessions alive based on central INs
• Mobility • LAN-MAN: Cellular IP
– Within subnet • Wide Area: Mobile IP
– Within domain
– Global • Universal (any net): SIP
• AAA and NAIs
• Location privacy
• QoS for r.t. communications

38
 Presence, Instant Messaging and Voice

http://www.ietf.org/internet-drafts/draft-ietf-impp-model-03.txt
39
 IP SIP Phones and Adaptors
Are Internet hosts
• Choice of application
• Choice of server 1
• IP appliance

 
Implementations
2
• 3Com (2)                
• Cisco
• Columbia University
• Mediatrix (1)
3
• Nortel (3)
• Pingtel
40
 SIP 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)
41
 References

Book on “Internetworking Multimedia” by Jon Crowcroft, Mark Handley, Ian Wakeman,

UCL Press, 1999 by Morgan Kaufman (USA) and Taylor Francis (UK)

RFC 2543: “SIP: Session Initiation Protocol”

ftp://ftp.isi.edu/in-notes/rfc2543.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/~hgs/sip/papers.html

42
 Relevant IETF Working Groups
http://ietf.org/html.charters/wg-dir.html

• 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)

This large work effort may cause the complete re-engineering

of communication systems and services
43