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

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

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)

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 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)

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 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.

Protocols We Have Signaling Media Transport Supporting 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 SIP SIP TCP or UDP TCP or UDP IP IP Lower layer Internet

Other Protocols

Proxy and Redirect Servers

SIP Methods INVITE BYE OPTIONS ACK REGISTER CANCEL

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”

Message Example: INVITE First 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 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

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 “Contact 1234@10.1.1.1, 1234@10.1.1.2 and 1234@10.1.1.3” Location Database INVITE sip:1234@10.1.1.3 “Where is sip:1-800-GO-CISCO@cisco.com?” INVITE sip:1234@10.1.1.2 Proxy / Redirect Server INVITE sip:1234@10.1.1.1 INVITE sip:1-800-GO-CISCO@cisco.com 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 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) At Columbia 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 Call from Jack When bob is at his office in Columbia, Jack initialize a call to bob placed to bob@lucent.com 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 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)

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

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 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.

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 Features distribution Security Old feature interaction New feature interaction Features distribution Inside end device or on internet Security Packets go through public Internet

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)

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