Internet Telephony based on SIP

SMU - Dallas April 28, May 1, 2000 Henry Sinnreich, MCI WorldCom Alan Johnston, MCI WorldCom

Internet Multimedia
Real Time Protocol (RTP) media packets Real Time Control Protocol (RTCP) monitor & report Session Announcement Protocol (SAP) Session Description Protocol (SDP) Session Initiation Protocol (SIP) Real Time Stream Protocol (RTSP) play out control Synchronized Multimedia Integration Language (SMIL) mixes audio/video with text and graphics

References: Search keyword at http://www.rfc-editor.org/rfc.html

For SMIL - http://www.w3.org/AudioVideo/
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Telephony on the Internet

may not be a stand-alone business, but part of IP services

SIP/RTP Media Architecture

Telephone Gateway SIP client
CAS, Q.931, SS7

SG
PCM

MG

MGCP

Public IP Backbone Goes everywhere End-to-end control Consistent for all services DNS mobility Messaging SIP Web Directory RTP Security QoS Media services Sessions Telephony

Any other sessions

Commercial Grade IP Telephony

Assure baseline PSTN features Leverage and Commonality of telephony with the Web/Internet New services (new revenue) Scalability (Web-like) Baseline PSTN&PBX features

Client & user authentication

Accounting assured QoS QoS assured signaling Security assured signaling Hiding of caller ID & location

Better than PSTN features New & fast service creation Internet (rapid) scalability Mobility Dynamic user preferences End-to-end control Service selection Feature control Mid-call control features Pre-call Mid-call
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Internet End-to-End Control

No single point of failure All services enabled by protocols: From ftp to web Elective Server Elective Server User has control of all applications and choice of servers

USER

Internet
R R

USER

Dumb Network
Services supported by interfaces and central controllers

ITU Intelligent Network Control: POTS, ISDN, BISDN, FR, ATM, H.323, MEGACO/H.248, GSM
Central Control
SW
NNI

Central Control

Central Control
SW NNI SW UNI

User has little control

USER
UNI

SW

SW

SW

USER
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SIP vs. flavors of IPDC, SGSP, MGCP, MEGACO, H.248 (Internet Client-Server vs. Telco Master-Slave Protocols) CAS, Q.931, SS7 GC MCGP MG SIP, H.323

PSTN
Legend CG: gateway Controller MG: Media Gateway

Internet
RTP

PCM

Absorbs PSTN complexity at the edge of IP

1. IP Telephony Gateway

GC PSTN MG MCGP MG PSTN RG

GC MCGP Internet

IP
2. Softswitch a la IN

TR 303
phone to phone only PSTN services single vendor solution

3. Residential GWY

breaks e-2-e control model no services integration no choice of server and apps unequal access is reinvented

IP Communications
Complete integration of all services under full user control Web-like: PSTN/PBX-like: Presence POTS Voice and text chat AIN CS-1, CS-2 Messaging PBX & Centrex Voice, data, video User has control of: Multiparty All addressable devices Conferencing Caller and called party Education Games preferences Any quality Better quality than 3.1 kHz Most yet to be invented Mixt Internet-PSTN: ClicknConnect, ICW, unified messaging
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Development of SIP

IETF - Internet Engineering Task Force

MMUSIC - Multiparty Multimedia Session Control Working Group SIP developed by Handley, Schulzrinne, Schooler, and Rosenberg

Submitted as Internet-Draft 7/97

Assigned RFC 2543 in 3/99 Internet Multimedia Conferencing Architecture. H.323 used for IP Telephony since 1994 Problems: No new services, addressing, features Concerns: scalability, extensibility

Alternative to ITUs H.323

SIP Philosophy

Internet Standard

IETF - http://www.ietf.org Utilizes rich Internet feature set Text based TCP, UDP, X.25, frame, ATM, etc. Support of multicast

Reuse Internet addressing (URLs, DNS, proxies)

Reuse HTTP coding

Makes no assumptions about underlying protocol:

SIP Clients and Servers - 1

SIP uses client/server architecture Elements:

SIP User Agents (SIP Phones) SIP Servers (Proxy or Redirect - used to locate SIP users or to forward messages.)

Can be stateless or stateful To PSTN for telephony interworking To H.323 for IP Telephony interworking

SIP Gateways:

Client - originates message Server - responds to or forwards message

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SIP Clients and Servers - 2

Logical SIP entities are:

User Agents

User Agent Client (UAC): Initiates SIP requests User Agent Server (UAS): Returns SIP responses Registrar: Accepts REGISTER requests from clients Proxy: Decides next hop and forwards request Redirect: Sends address of next hop back to client

Network Servers

The different network server types may be collocated

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SIP Addressing
Uses Internet URLs
Uniform Resource Locators

Supports both Internet and PSTN addresses

General form is name@domain To complete a call, needs to be resolved down to User@Host Examples:
sip:[email protected] sip:J.T. Kirk <[email protected]> sip:[email protected];user=phone sip:[email protected] sip:[email protected];phone-context=VNET
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SIP Session Setup Example

SIP User Agent Client SIP User Agent Server

INVITE sip:[email protected]

200 OK ACK Media Stream BYE 200 OK host.wcom.com sip.uunet.com

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Proxy Server Example

SIP User Agent Client
INVITE sip:[email protected]

SIP Proxy Server

SIP User Agent Server

INVITE sip:[email protected]

200 OK

200 OK

ACK

Media Stream
BYE 200 OK

host.wcom.com

server.wcom.com

sip.uunet.com

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Redirect Server Example

SIP User Agent Client SIP Redirect Server
REGISTER [email protected] INVITE sip:[email protected] 302 Moved sip:[email protected]

SIP User Agent Server 200 OK

ACK

RS
2

C
INVITE sip:[email protected]

UAS

180 Ringing 200 OK ACK

Media Stream
host.wcom.com server.wcom.com sip.uunet.com

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SIP Requests
SIP Requests (Messages) defined as:
Method SP Request-URI SP SIP-Version CRLF
CRLF=Carriage Return and Line Feed) (SP=Space,

Example: INVITE sip:[email protected] SIP/2.0

Method Description A session is being requested to be setup using a specified media

INVITE ACK

Message from client to indicate that a successful response to an INVITE has been received

OPTIONS A Query to a server about its capabilities BYE CANCEL

A call is being released by either party

Cancels any pending requests. Usually sent to a Proxy Server to cancel searches

REGISTER Used by client to register a particular address with the SIP server

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SIP Requests Example

Required Headers (fields):
INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP host.wcom.com:5060 From: Alan Johnston <sip:[email protected]> To: Jean Luc Picard <sip:[email protected]> Call-ID: [email protected] CSeq: 1 INVITE

Uniquely identify this session request

Via: Shows route taken by request. Call-ID: unique identifier generated by client. CSeq: Command Sequence number

generated by client Incremented for each successive request

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SIP Requests Example

Typical SIP Request:
INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP host.wcom.com:5060 From: Alan Johnston <sip:[email protected]> To: Jean Luc Picard <sip:[email protected]> Call-ID: [email protected] CSeq: 1 INVITE Contact: sip:[email protected] Subject: Where are you these days? Content-Type: application/sdp Content-Length: 124 v=0 o=ajohnston 5462346 332134 IN IP4 host.wcom.com s=Let's Talk t=0 0 c=IN IP4 10.64.1.1 m=audio 49170 RTP/AVP 0 3
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SIP Responses
SIP Responses defined as (HTTP-style):
SIP-Version SP Status-Code SP Reason-Phrase CRLF
(SP=Space, CRLF=Carriage Return and Line Feed)

Example: SIP/2.0 404 Not Found First digit gives Class of response:
Description 1xx 2xx 3xx 4xx 5xx 6xx Informational Request received, continuing to process request. Success Action was successfully received, understood and accepted. Redirection Further action needs to be taken in order to complete the request. Client Error Request contains bad syntax or cannot be fulfilled at this server. Server Error Server failed to fulfill an apparently valid request. Global Failure Request is invalid at any server. Examples 180 Ringing 181 Call is Being Forwarded 200 OK 300 Multiple Choices 302 Moved Temporarily 401 Unauthorized 408 Request Timeout 503 Service Unavailable 505 Version Not Suported 600 Busy Everywhere 603 Decline

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SIP Responses Example

Required Headers:
SIP/2.0 200 OK Via: SIP/2.0/UDP host.wcom.com:5060 From: Alan Johnston <sip:[email protected]> To: Jean Luc Picard <sip:[email protected]> Call-ID: [email protected] CSeq: 1 INVITE

Via, From, To, Call-ID, and CSeq are copied exactly from Request. To and From are NOT swapped!

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SIP Responses Example

Typical SIP Response
(containing SDP)
SIP/2.0 200 OK Via: SIP/2.0/UDP host.wcom.com From: Alan Johnston <sip:[email protected]> To: Jean Luc Picard <sip:[email protected]> Call-ID: [email protected] CSeq: 1 INVITE Contact: sip:[email protected] Subject: Where are you these days? Content-Type: application/sdp Content-Length: 107 v=0 o=picard 124333 67895 IN IP4 uunet.com s=Engage! t=0 0 c=IN IP4 11.234.2.1 m=audio 3456 RTP/AVP 0

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Forking Proxy Example

SIP User Agent Client INVITE sip:[email protected] 100 Trying SIP Proxy Server SIP User Agent Server 1 INVITE INVITE 404 Not Found ACK 180 Ringing 200 OK 200 OK SIP User Agent Server 2

S1 C S2

Fork
180 Ringing

ACK Media Stream BYE 200 OK

host.wcom.com

proxy.wcom.com

sip.mci.com

sip.uunet.com

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SIP Headers - Partial List

Header Accept Authorization Call-ID Description Indicates acceptable formats. Contains encryption information Used to uniquely identify a particular session or registration messages. Should have randomness to ensure overall global uniqueness. Alternative SIP URL for more direct message routing. Octet count in message body. Content type of message body Command Sequence number used to distinguish different requests during the same session. Encryption information. Used to indicate when the message content is no longer valid. Can be a number of seconds or a date and time. Accept: application/sdp Accept: currency/dollars Authorization: pgp info Call-ID: [email protected] Call-ID: Jan-01-1999-1510i: [email protected] [email protected] Examples

Contact Content-Length Content-Type CSeq

Contact: W. Riker, Acting Captain <[email protected]> Contact: [email protected]; expires=3600 m: [email protected] Content-Length: 285 Content-Type: application/sdp c: application/h.323 CSeq: 1 INVITE CSeq: 1000 INVITE CSeq: 4325 BYE CSeq: 1 REGISTER Encryption: pgp info Expires: 60 Expires: Thu, 07 Jan 1999 17:00 CST

Encryption Expires

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SIP Headers - Continued

From Required field containing the originating SIP URL. Can also include a display name. From: Dana Scully <sip:[email protected]> From: sip:[email protected]; tag=1234567 f: sip: [email protected] Max-Forwards: 10

Max-Forwards Count decremented by each server forwarding the message. When goes to zero, server sends a 483 Too Many Hops response. Priority Can specify message priority

Record-Route Added to a request by a proxy that needs to be in the path of future messages. Require Indicates options necessary for the session.

Priority: normal Priority: emergency Record Route: sip.mci.com Require: local.telephony Response-Key: pgp info

Response-Key Contains PGP key for encrypted response expected. Retry-After

Retry-After: 3600 Indicates when the resource may be available. Can be a number of seconds or a Retry-After: Sat, 01 Jan 2000 00:01 GMT date and time.

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SIP Headers - Continued

Route Subject To Determines the route taken by a message. Can be used to indicate nature of call. Route: orinoco.brooks.net Subject: More about SIP s: Youd better answer!

Unsupported Via Warning

To: Fox Mulder Required field containing the recipient SIP URL. May <sip:[email protected]> To: sip:[email protected]; contain a display name. tag=314 t: sip:[email protected]; tag=52 Unsupported: tcap.telephony Lists features not supported by server. Via: SIP/2.0/UDP sip.mfs.com Via: SIP/2.0/TCP uunet.com v: SIP/2.0/UDP 192.168.1.1 Contains a code and text to Warning: 331 Unicast not available warn about a problem Used to show the path taken by the request.

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Via Headers and Routing

Via headers are used for routing SIP messages Requests

Request initiator puts address in Via header Servers check Via with senders address, then add own address, then forward. (if different, add received parameter)

Responses

Response initiator copies request Via headers. Servers check Via with own address, then forward to next Via address
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SIP Firewall Considerations

Firewall Problem

Can block SIP packets Can change IP addresses of packets

TCP can be used instead of UDP Record-Route can be used:

ensures Firewall proxy stays in path Clients and Servers copy Record-Route and put in Route header for all messages

A Firewall proxy adds Record-Route header

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SIP Message Body

Message body can be any protocol Most implementations:

SDP - Session Description Protocol RFC 2327 4/98 by Handley and Jacobson

http://www.ietf.org/rfc/rfc2327.txt

Used to specify info about a multi-media session. SDP fields have a required order For RTP - Real Time Protocol Sessions:

RTP Audio/Video Profile (RTP/AVP) payload descriptions are often used

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SDP Examples
SDP Example 1
v=0 o=ajohnston +1-613-555-1212 IN IP4 host.wcom.com s=Let's Talk Field t=0 0 Version c=IN IP4 101.64.4.1 m=audio 49170 RTP/AVP 0 3
Origin

Descripton v=0 o=<username> <session id> <version> <network type> <address type> <address> s=<session name> t=<start time> <stop time> c=<network type> <address type> <connection address> m=<media> <port> <transport> <media format list>

SDP Example 2
v=0 o=picard 124333 67895 IN IP4 uunet.com s=Engage! t=0 0 c=IN IP4 101.234.2.1 m=audio 3456 RTP/AVP 0

Session Name Times Connection Data Media

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Another SDP Example

v=0 o=alan +1-613-1212 IN host.wcom.com s=SSE University Seminar - SIP i=Audio, Listen only u=http://sse.mcit.com/university/ [email protected] p=+1-329-342-7360 c=IN IP4 10.64.5.246 b=CT:128 t=2876565 2876599 m=audio 3456 RTP/AVP 0 3 a=type:recvonly

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Authentication & Encryption

SIP supports a variety of approaches:

end to end encryption hop by hop encryption Responds to INVITEs with 407 ProxyAuthentication Required Client re-INVITEs with Proxy-Authorization header. Responds to INVITEs with 401 Unathorized Client re-INVITEs with Authorization header
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Proxies can require authentication:

SIP Users can require authentication:

SIP Encryption Example

INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP host.wcom.com:5060 From: Alan Johnston <sip:[email protected]> To: Jean Luc Picard <sip:[email protected]> Call-ID: [email protected] CSeq: 1 INVITE Content-Length: 224 Encryption: PGP version=2.6.2, encoding=ascii q4aspdoCjh32a1@WoiLuaE6erIgnqD3erDg8aFs8od7idf@ hWjasGdg,ddgg+fdgf_ggEO;ALewAKFeJqAFSeDlkjhasdf kj!aJsdfasdfKlfghgasdfasdfa|Gsdf>a!sdasdf3w2945 1k45mser?we5y;343.4kfj2ui2S8~&djGO4kP%Hk#(Khuje fjnjmbm.sd;dal;12;123=]aw;erwAo3529ofgk
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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
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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 Dont 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
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SIP User Location Example

SIP supports mobility across networks and devices Q=quality gives preference
SIP/2.0 302 Moved temporarily Contact: sip:[email protected] ;service=IP,voice mail ;media=audio ;duplex=full ;q=0.7 Contact : phone: +1-972-555-1212; service=ISDN ;mobility=fixed; language=en,es, ;q=0.5 Contact : phone: +1-214-555-1212; service=pager ;mobility=mobile ;duplex=send-only ;media=text; q=0.1; priority=urgent ;description=For emergency only Contact : mailto: [email protected]
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SIP Mobility Support

4 Mobile Host 5 SIP Redirect Server

Foreign Network 7

SIP Proxy Server

Home Network

6 Global: Wire and wireless No tunneling required No change to routing For fast hand-offs use: Use Cellular IP or Use DRCP
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1 INVITE

2 302 moved temporarily

Corresponding Host

3, 4 INVITE
5, 6 OK 7 Data

SIP Mobility Pre-call mobility

Mid-call mobility

MH can find SIP server via multicast REGISTER MH acquires IP address via DHCP MH updates home SIP server

MH->CH: New INVITE with Contact and updated SDP Re-registers with home registrar

Need not bother home registrar: Use multi-stage registration

Recovery from disconnects
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Mobile IP Communications
Mobile IP Requirements Transparency above L2: Evolution of Wireless Mobility

Move but keep IP address and all sessions alive

Circuit Switched Mobility

based on central INs

Mobility

LAN-MAN:

Cellular IP

Within subnet Within domain Global

Wide Area: Mobile IP Universal (any net): SIP

AAA and NAIs Location privacy QoS for r.t. communications

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Presence, Instant Messaging and Voice

http://www.ietf.org/internet-drafts/draft-ietf-impp-model-03.txt
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IP SIP Phones and Adaptors

Are Internet hosts Choice of application

Choice of server
IP appliance Implementations

3Com (2)
Cisco

Columbia University

Mediatrix (1)
Nortel (3) Pingtel
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SIP Summary

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)
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References
Book on Internetworking Multimedia by Jon Crowcroft, Mark Handley, Ian Wakeman, UCL Press, 1999 by Morgan Kaufman (USA) and Taylor Francis (UK) RFC 2543: SIP: Session Initiation Protocol ftp://ftp.isi.edu/in-notes/rfc2543.txt

The IETF SIP Working Group home page http://www.ietf.org/html.charters/sip-charter.html

SIP Home Page http://www.cs.columbia.edu/~hgs/sip/ Papers on IP Telephony http://www.cs.columbia.edu/~hgs/sip/papers.html
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Relevant IETF Working Groups

http://ietf.org/html.charters/wg-dir.html

Audio/Video Transport (avt) - RTP Differentiated Services (diffserv) QoS in backbone IP Telephony (iptel) CPL, GW location, TRIP Integrated Services (intserv) end-to-end QoS Media Gateway Control (megaco) IP telephony gateways Multiparty Multimedia Session Control (mmusic) SIP, SDP, conferencing PSTN and Internet Internetworking (pint) mixt services Resource Reservation Setup Protocol (rsvp) Service in the PSTN/IN Requesting InTernet Service (spirits) Session Initiation Protocol (sip) signaling for call setup Signaling Transport (sigtran) PSTN signaling over IP Telephone Number Mapping (enum) surprises ! Instant Messaging and Presence Protocol (impp)

This large work effort may cause the complete re-engineering of communication systems and services
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