Transcript jiang

Telephony Features with SIP
DongMei Jiang
Yong He
March 24, 2002
Contents
Introduction
 Internet telephony
 SIP telephony features
 Case studies
 Pros and Cons
 Conclusion
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Features and Services
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Features
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“management-based capabilities which a a unit
of one or more telecommunications or
telecommunications network provides to a user”
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Services
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A set of features (not a very clear distinction)
Telephony features history
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In-band signaling
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Out-of-band signaling
Intelligent networks
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only dial and receive calls
800 service , call forward, three way calling
Voice over IP
Internet telephony

Wide range, flexible and new features such as
Caller selection etc.
Feature Classification

Basic Features
(unit to provide base capabilities
to a user)
Network features (supported by network)
 Client Features (depend on end devices or

stream contents)
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Bundle Features
(package of basic features)
Traditional Features
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(ITU-T) Descriptions of features
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New features
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Q.1211: Introduction to Intelligent Network CS1 (CW)
Q1221: Introduction to Intelligent Network CS2 (SCF)
wireless services, multimedia services and service
management services etc.
No standard specifications for features

One feature may have different name (ex. CFU and CF)
Internet Telephony

Internet telephony is all about IP
Runs on top of IP and utilizes the IP
service model.
 It is not about re-engineering PSTN -PSTN is good enough!

Calls over the Internet
PC-to-PC
 PC-to-Phone
 Phone-to-PC
 Phone-to-Phone
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Protocols Needed
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Signaling Protocol
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Media Transport Protocol
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locate users, set up, modify and tear
down sessions
transmission of packetized audio/video
Supporting Protocol
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Gateway location, QoS, address
translation,etc.
Protocols We Have
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Signaling
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Media
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RTP
Transport
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SIP (IETF), H.323 (ITU-T)
TCP, UDP
Supporting
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DNS, RSVP, TRIP, etc
What is SIP?
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Session Initiation Protocol
Defined in FRC2543 (March 1999).
“… is an application-layer control protocol
that can establish, modify and terminate
multimedia sessions or calls.”
Modeled after protocols SMTP and HTTP
One of the protocols supporting Internet
Telephony
End-to-end, client/server
General Purpose Protocol
SIP is NOT transport protocol
 SIP is not limited to Internet
telephony
 Arbitrary services could be built on
top of SIP.

SIP Placement
SIP
SIP
TCP or UDP
TCP or UDP
IP
IP
Lower layer
Lower layer
Internet
Other Protocols
Proxy and Redirect Servers
SIP Methods

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INVITE
BYE
OPTIONS
ACK
REGISTER
CANCEL
Message Structure
First Line
Headers
METHOD “URL” “SIP version”
Via: “URL”
From: “URL”
To: “URL”
Call-ID: “URL”
Cseq: 1 INVITE
Contact: “URL”
Expires: “time”
Message Body
Via: “URL”
Subject: “Description of subject “
Call-ID: “an IP Address”
Content-Endcoding: “Appropriate Information”
Message Example: INVITE
First line
INVITE sip: [email protected] SIP/2.0
Headers
Via: SIP/2.0/UDP lucent.com: 4545
From: User A <sip: [email protected]>
To: User B <sip:[email protected]>
Call-ID: [email protected]
Cseq: 1 INVITE
Subject: test SIP message
Contact: User B <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 187
Message Body
v=0
o=user1 53655765 2353687637 IN IP4 128.3.4.5
c=IN IP4 224.2.0.1/127
t=0 0
m=audio 3456 RTP/AVP 0
SIP Response Codes
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Borrowed from HTTP.
1xx
 2xx
 3xx
 4xx
 5xx
 6xx
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Informational
Success
Redirection
Client Error
Server Failure
Global Failure
SIP Functions

Name translation and user location
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Feature negotiation
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Allows a group of participants to negotiate on the
media exchanged and parameters preferred
Call participant management
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Mapping names to identify a callee and the eventual
location
It may be depend on caller and callee preferences
In the course of a call, media session composition is
still adjustable when necessary
Call feature changes

Can adjust the session composition in the session
processing
Telephony features with SIP

Solve some existing problems in
PSTN
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Signal overloading etc.
Wide range, high flexibility of
services
Take over PSTN telephony features
 Enhance PSTN telephony features
 Introduce new telephony features not
realizable in PSTN
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Low cost
Some new Features with SIP
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Integration of data, voice and fax
Sound grading
Video telephony
Unified messaging
A virtual second line
Web-based call centers
Low-cost voice calls
Real-time billing
Remote teleworking
Enhanced teleconferencing
PSTN Features with SIP
Features Implemented by SIP Phone
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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
PSTN Features with SIP
Features Implemented by SIP Server
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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
Personal Mobility
Personal mobility v.s. terminal
mobility
 Person uses different Devices and
possibly address
 REGISTER binds a person to a
device
 Proxy and redirect translate address
to location and device
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SIP For Presence
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Instant messaging (IM) and presence based
services, offered by AOL, Yahoo! and MSN,
nearly 100 million users.
Proprietary technology, with no technical
standard to support interoperability.
SIP extension, SIP for Instant Messaging
and Presence Leveraging Extensions
(SIMPLE)
SIMPLE is built in Microsoft Windows XP.
AOL has committed to using SIMPLE.
Case Study 1:
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Scenario
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successful call A to B
B put A on hold
B returns to A
Simple Call Hold
Case study 2:
Call Forward Unconditionally
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Scenario
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A calls to B
The call is
forward to C
A talks to C
Case Study 3: Call Forking
“Contact [email protected],
[email protected] and [email protected]”
Location Database
“Where is sip:[email protected]?”
INVITE sip:[email protected]
Proxy / Redirect
Server
INVITE sip:[email protected]
LOCAL PSTN
Forked Calls can be in parallel or sequential. The first phone to answer
will get the call, the others will get a CANCEL from the Proxy Server.
Case study 4:
Home Phone
Home Phone Scenario
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One caller sends a SIP INVITE to
[email protected](1)
the internet service provider (ISP) consults its
database(2), the proxy server forks and sends
out three INVITE requests to family member1, 2
and 3 (3, 4, 5).
When first member phone is picked up(6), all
other phones are not ringing anymore (7, 8).
Server forwards call acceptance back to caller(9).
When one member is talking on the phone,
other member can also join the talk by picking
up their phones (10).
Case study 5:
Personal Mobility
Personal Mobility Scenario
Bob has
• a single published IP telephony phone address: [email protected] is registered in
Lucent SIP server and an office (at Lucent Technologies location)
• a lab and an office (Columbia University)
At Columbia
• register Lucent SIP server with his Columbia address
[email protected] as a forwarding address (1)
• registers the lab machine [email protected] and the office machine
[email protected] with the Columbia SIP server (2, 3).
• Set his lab’s computer forward calls to his Lucent address
Call from Jack
• When bob is at his office in Columbia, Jack initialize a call to bob
placed to [email protected] at Lucent Technologies location (4).
Personal Mobility Scenario
(cont’n)
•
•
•
•
•
The server checks its registration and policy in database and decides to forward the
request to [email protected] by looking up columbia.edu in Name Domain System
(DNS) and get the main Columbia SIP server address (5, 6).
Columbia server find [email protected] in database and two end devices listed
under the address, forks and sends a call request to lab and office machine (7, 8, 9)
cause office phone to ring.
Lab phone sends request to Lucent server by its previous configuration (10). Using an
loop detection capability in SIP, Lucent server detected the loop error occurred and
send error response back to lab machine (11). In turn, returns an error code to the
Columbia server (12)
Bob answer the phone call in the office, sending an acceptance response back to the
Columbia server (13). Received both response back, the server forwards the call
acceptance back to Lucent server (14), which forwards the request back to the
original caller, Jack (15). All Sip session states in both server can be destroyed now.
Call setup and processed by intermediate servers between Jack and Bob (16)
Case study 6:
Configuration:

Caller phone destination
for the address
[email protected]
to a particular multicast
address

S1, S2, S3 listen for
calls request to on
this address
Caller Selection
Caller Selection Scenario
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Caller send message to [email protected] multicast
address, all S1, S2 and S3 get the INVITE request (1)
S1 answers first with response multicast. Like CANCEL, S2 and
S3 phones stop ring. Call is established between caller and S1
(2)
S2 join the answer session with his/her acceptance is also
multicast (3)
Received S2 acceptance, the caller can take any an action
•
•
•
•
Accept both S1 and S2 to a multicast media conference
Accept one and hang up anther one
Hang up both S1 and S2
Accept S1 and redirect S2 to a voice mail
Case Study 7: Sipc 1.72
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•
SIP User Agent
Sipc 1.72 :
Incoming call window
Sipc 1.72 Overview
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sipc is a SIP user agent that can be used for Internet telephony
calls, multimedia conferences, instant messaging, web browsing
sharing and device control. It supports a range of media types,
such as audio, video, text and white board, and can be extended
easily to additional media types.
sipc can communicate with SIP redirect, proxy and registration
servers such as sipd and other SIP user agents. It includes a
user agent client which can send requests to SIP servers and a
user agent server which handles incoming calls.
sipc runs on a range of platforms: Windows 95/98/NT/2000/XP,
Linux and Solaris.
sipc does not provide audio and video functionality itself; rather,
it uses external media application for handling media streams.
Currently, it uses rat (Robust Audio Tool) as its audio application
for both Unix and Windows version, vic as the video application,
wb (for Unix) and wbd (for Windows) as white board application.
Key Benefits with SIP
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Simplicity
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Extensibility
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Built in a rich set of extensibility and compatibility functions
by learning lessons from HTTP and SMTP
Modularity
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Only 99 page long specification, 42 headers
SIP message encoded as text, parsing and generation are
simple
Call signaling, user location, basic registration reside in SIP
Other functions such as QOS, session content description etc.
are orthogonal and reside in different protocols
Integration
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HTTP, SMTP, RTSP etc.
Problems and Difficulties
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Potential problems
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QoS challenges
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Private address passing firewall and accepted by internet
Discussed at internet conference, Birds of A Feature session
Unlike PSTN, a circuit-switched network, IP telephone QoS faces
technical challenges such as loss, delay, and jitter.
New protocols and techniques need to be incorporated.
(being carried out by the Differentiated Service and IP telephony
groups of IETF)
Many effect factors
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Features existed in PSTN
Non architecture
New feature issues (standards etc.)
Feature distribution and interaction
Other Concerns
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Feature interaction
Old feature interaction
 New feature interaction
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Features distribution
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Inside end device or on internet
Security
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Packets go through public Internet
Conclusion
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SIP is:
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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:
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Going to make PSTN interworking easy
Going to solve all IP Telephony issues (QoS)
Conclusion
(cont’n)
SIP, next generation telephony
signaling protocol
 Internet telephony with SIP provides
wealthy telephony features with low
price
 It is a long way to go to realize the
next generation telephony, an
common application over internet
