Telephony Features with SIP

DongMei JiangYong He

March 24, 2002

Contents

Introduction Internet telephony SIP telephony features Case studies Pros and Cons Conclusion

Features and Services Features

“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)

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.

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)

Traditional Features (ITU-T) Descriptions of 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)

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

Protocols Needed Signaling 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.

Protocols We Have Signaling

SIP (IETF), H.323 (ITU-T) Media

RTP Transport

TCP, UDP Supporting

DNS, RSVP, TRIP, etc

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

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

TCP or UDP

IP

TCP or UDP

SIP SIP

Lower layer

IP

Lower layer

Internet

Other Protocols

Proxy and Redirect Servers

SIP Methods INVITE BYE OPTIONS ACK REGISTER CANCEL

Message StructureFirst Line METHOD “URL” “SIP version”

Headers 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: INVITEFirst line INVITE sip: uB@lucent.com SIP/2.0

Headers Via: SIP/2.0/UDP lucent.com: 4545 From: User A <sip: uA@lucent.com> To: User B <sip:uB@site.uottawa.ca> Call-ID: 34567@lucent.com Cseq: 1 INVITE Subject: test SIP message Contact: User B <sip:uB@cs.site.uottawa.ca> 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 Borrowed from HTTP.

1xx Informational 2xx Success 3xx Redirection 4xx Client Error 5xx Server Failure 6xx Global Failure

SIP Functions Name translation and user location

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

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

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

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

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

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

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.

Case Study 1: Simple Call Hold

Scenario successful call A to B B put A on hold B returns to A

Case study 2: Call Forward Unconditionally

Scenario A calls to B The call is

forward to C A talks to C

Case Study 3: Call Forking

LOCAL PSTN

Proxy / Redirect Server

Location Database

INVITE sip:1-800-GO-CISCO@cisco.com

“Where is sip:1-800-GO-CISCO@cisco.com?”

“Contact 1234@10.1.1.1, 1234@10.1.1.2 and 1234@10.1.1.3”

INVITE sip:1234@10.1.1.1

INVITE sip:1234@10.1.1.2

INVITE sip:1234@10.1.1.3

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.

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 One caller sends a SIP INVITE to

smith_family@isp.com(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: bob@lucent.com is registered in Lucent SIP server and an office (at Lucent Technologies location) • a lab and an office (Columbia University)

• register Lucent SIP server with his Columbia address bob@columbia.edu as a forwarding address (1)• registers the lab machine bob@lab.columbia.edu and the office machine bob@office.columbia.edu with the Columbia SIP server (2, 3).• Set his lab’s computer forward calls to his Lucent address

• When bob is at his office in Columbia, Jack initialize a call to bob placed to bob@lucent.com at Lucent Technologies location (4).

At Columbia

Call from Jack

Personal Mobility Scenario (cont’n)

• The server checks its registration and policy in database and decides to forward the request to bob@columbia.edu by looking up columbia.edu in Name Domain System (DNS) and get the main Columbia SIP server address (5, 6).

• Columbia server find Bob@columbia.edu 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)

Configuration:

Caller phone destinationfor the address sysadmins@company.com to a particular multicast address

S1, S2, S3 listen for

calls request to on this address

Case study 6: Caller Selection

Caller Selection Scenario Caller send message to sysadmins@company.com 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

SIP User Agent

•

•

Sipc 1.72 : Incoming call window

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.

Key Benefits with SIP Simplicity

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.

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

Other Concerns Feature interaction

Old feature interaction New feature interaction

Features distribution Inside end device or on internet

Security Packets go through public Internet

Conclusion 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)

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