1. Telephony Features with SIP
DongMei Jiang
Yong He
March 24, 2002

2. Contents Introduction Internet telephony SIP telephony features
Case studies
Pros and Cons
Conclusion

3. Features and Services Features Services
“management-based capabilities which a a unit of one or more telecommunications or telecommunications network provides to a user”
Services
A set of features (not a very clear distinction)

4. Telephony features history
In-band signaling
only dial and receive calls
Out-of-band signaling
Intelligent networks
800 service , call forward, three way calling
Voice over IP
Internet telephony
Wide range, flexible and new features such as Caller selection etc.

5. 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)
Bundle Features (package of basic features)

6. Traditional Features (ITU-T) Descriptions of features New features
Q.1211: Introduction to Intelligent Network CS1 (CW)
Q1221: Introduction to Intelligent Network CS2 (SCF)
New features
wireless services, multimedia services and service management services etc.
No standard specifications for features
One feature may have different name (ex. CFU and CF)

7. 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!

8. Calls over the Internet
PC-to-PC
PC-to-Phone
Phone-to-PC
Phone-to-Phone

9. Protocols Needed Signaling Protocol Media Transport Protocol
locate users, set up, modify and tear down sessions
Media Transport Protocol
transmission of packetized audio/video
Supporting Protocol
Gateway location, QoS, address translation,etc.

10. Protocols We Have Signaling Media Transport Supporting
SIP (IETF), H.323 (ITU-T)
Media
RTP
Transport
TCP, UDP
Supporting
DNS, RSVP, TRIP, etc

11. What is SIP? 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

12. General Purpose Protocol
SIP is NOT transport protocol
SIP is not limited to Internet telephony
Arbitrary services could be built on top of SIP.

13. SIP Placement SIP SIP TCP or UDP TCP or UDP IP IP Lower layer
Internet

14. Other Protocols

15. Proxy and Redirect Servers

16. SIP Methods
INVITE
BYE
OPTIONS
ACK
REGISTER
CANCEL

17. Message Structure First Line METHOD “URL” “SIP version” Headers
Via: “URL”
From: “URL”
To: “URL”
Call-ID: “URL”
Cseq: 1 INVITE
Contact: “URL”
Expires: “time”
Message Body
Subject: “Description of subject “
Call-ID: “an IP Address”
Content-Endcoding: “Appropriate Information”

18. Message Example: INVITE
First line INVITE sip: SIP/2.0
Headers Via: SIP/2.0/UDP lucent.com: 4545
From: User A <sip:
To: User B
Call-ID:
Cseq: 1 INVITE
Subject: test SIP message
Contact: User B
Content-Type: application/sdp
Content-Length: 187
Message Body v=0
o=user IN IP
c=IN IP /127
t=0 0
m=audio 3456 RTP/AVP 0

19. SIP Response Codes Borrowed from HTTP. 1xx Informational 2xx Success
3xx Redirection
4xx Client Error
5xx Server Failure
6xx Global Failure

20. SIP Functions Name translation and user location Feature negotiation
Mapping names to identify a callee and the eventual location
It may be depend on caller and callee preferences
Feature negotiation
Allows a group of participants to negotiate on the media exchanged and parameters preferred
Call participant management
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

21. Telephony features with SIP
Solve some existing problems in PSTN
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
Low cost

22. Some new Features with SIP
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

23. 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 =
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

24. 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
Voic server 302 Moved Temporarily response with Contact of Voic Server
Follow Me Service: Use forking proxy to try multiple locations at the same time
Caller-ID blocking - Privacy: Server encrypts From information

25. 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

26. SIP For Presence
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.

27. Case Study 1: Simple Call Hold
Scenario
successful call A to B
B put A on hold
B returns to A

28. Case study 2: Call Forward Unconditionally
Scenario
A calls to B
The call is
forward to C
A talks to C

29. Case Study 3: Call Forking
“Contact
and
Location Database
INVITE
“Where is
INVITE
Proxy / Redirect
Server
INVITE
INVITE
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.

30. Case study 4: Home Phone

31. Home Phone Scenario One caller sends a SIP INVITE to
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).

32. Case study 5: Personal Mobility

33. Personal Mobility Scenario
Bob has
a single published IP telephony phone address: 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
as a forwarding address (1)
registers the lab machine and the office machine
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 at Lucent Technologies location (4).

34. Personal Mobility Scenario (cont’n)
The server checks its registration and policy in database and decides to forward the request to by looking up columbia.edu in Name Domain System (DNS) and get the main Columbia SIP server address (5, 6).
Columbia server find 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)

35. Case study 6: Caller Selection
Configuration:
Caller phone destination
for the address
to a particular multicast
address
S1, S2, S3 listen for
calls request to on
this address

36. Caller Selection Scenario
Caller send message to 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

37. Case Study 7: Sipc 1.72
SIP User Agent
                                                                                                                                                                                              

38. Sipc 1.72 : Incoming call window

39. Sipc 1.72 Overview
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.

40. Key Benefits with SIP Simplicity Extensibility Modularity Integration
Only 99 page long specification, 42 headers
SIP message encoded as text, parsing and generation are simple
Extensibility
Built in a rich set of extensibility and compatibility functions by learning lessons from HTTP and SMTP
Modularity
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
HTTP, SMTP, RTSP etc.

41. Problems and Difficulties
Potential problems
Private address passing firewall and accepted by internet
Discussed at internet conference, Birds of A Feature session
QoS challenges
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
Features existed in PSTN
Non architecture
New feature issues (standards etc.)
Feature distribution and interaction

42. Other Concerns Feature interaction Features distribution Security
Old feature interaction
New feature interaction
Features distribution
Inside end device or on internet
Security
Packets go through public Internet

43. Conclusion SIP is: SIP is not: 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)

44. 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