telephony features with sip dongmei jiang yong he march 24, 2002
TRANSCRIPT
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: [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 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:[email protected]
“Where is sip:[email protected]?”
“Contact [email protected], [email protected] and [email protected]”
INVITE sip:[email protected]
INVITE sip:[email protected]
INVITE sip:[email protected]
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
[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)
• 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
• When bob is at his office in Columbia, Jack initialize a call to bob placed to [email protected] 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 [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)
Configuration:
Caller phone destinationfor the address [email protected] 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 [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
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