4 SIP Overview

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Session Initiation Protocol (SIP)

Transcript of 4 SIP Overview

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Session Initiation Protocol (SIP)

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Internet Telephony

SIP based VoIP Architecture

Legacy PBX

SIP User Agents (UA)

ApplicationServices

eMail LDAP Oracle XML

SIPSIP

RTP(Media)

SIP

CPL

CPL3pcc

PSTN

CAS or PRI

INTELLI

GENT

SERV

ICES

SIP Proxy, Registrar & Redirect Servers

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Internet Telephony

Basic SIP Call-Flow

SIP UA1 SIP UA2

INVITE w/ SDP for Media Negotiation

100 Trying

180/183 Ringing w/ SDP for MediaNegotiation

200 OK

200 OK

BYE

MEDIA

MEDIA

ACK

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Internet Telephony

SIP Call Flow with Proxy Server

Proxy Server

Register

OK (200)

Register

OK (200)

Invite InviteTrying (100) Ringing (180)Ringing (180)

OK (200)

OK (200)ACK ACK

RTP/RTCP mediachannels

User Agent 1 User Agent 1

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VoIP Migration

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Internet Telephony

Customer Premises

IP CoreNetwork

Step1: IPPBX deployments in Enterprises

DNS Server for URLresolutionLarge enterprises will handle VOIP callsdirectlyPSTN connectivity provided by MediaGatewaysRegulation can not stop spammersoutside USA

(similar to SMTP spam)

Customer Premises

PSTNNetwork

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Internet Telephony

Customer Premises

Carrier Network

Softswitches, MGWVoIP Proxy Server, SGW

SGC, VoIP CentrexServer,

Internet

STEP 2: H osted IP Centrex FW, NAT, VoIP service provided by Carrier Networks

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Internet Telephony

Customer Premises

Carrier Network

Step 3: Carrier VoIP Network

Softswitches, MGWVoIP Proxy Server, SGWSGC, VoIP CentrixServer,

Internet

- VoIP FW, NAT and Security provided byCarriers

VoIP Trunk

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SIP Architecture

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Internet Telephony

The Popularity of SIP

O riginally Developed in the MMUSICo A separate SIP working groupo RFC 2543, RFC 3261o Many developers

SIP + MGCP/MEGAC Oo The VoIP signaling in the future

³ back-off´ or SIPit (SIP Interoperability Tests)o Test products against each other o Will be hosted by ETSI

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Internet Telephony

SIP Architecture

A signaling protocolo The setup, modification, and tear-down of

multimedia sessionsSIP + SDP

o Describe the session characteristicsSeparate signaling and media streams

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Internet Telephony

SIP Network Entities

Clientso User agent clientso Application programs sending SIP requests

Serverso Responds to clients¶ requestsClients and servers may be in the same

platformProxy

o Acts as both clients and servers

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Internet Telephony

Four types of serverso Proxy servers

H andle requests or forward requests to other serversCan be used for call forwarding

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Redirect serverso Map the destination address to zero or more new addresseso Do not initiate any SIP requests

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Internet Telephony

A user agent server o Accept SIP requests and contacts the user o The user responds an SIP responseo A SIP deviceo E.g., an SIP-enabled telephone

A registrar o Accepts SIP REGISTER requestsIndicating the user is at a particular address

o Typically combined with a proxy or redirect server

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Internet Telephony

SIP Call Establishment

It is simpleo A number of interim responses

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Internet Telephony

SIP Advantages

Attempt to keep the signaling as simple as possible O ffer a great deal of flexibility

Various pieces of information can be included withinthe messages

o Including non-standard informationo Enable the users to make intelligent decisions

The user has control of call handlingo No need to subscribe call features

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Internet Telephony

Call Completion to Busy Subscriber service

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Internet Telephony

Via contains the address (e.g., pc33.atlanta.com)Contact contains a SIP or SIPS URI that represents a direct routeto contact the called party, usually composed of username at a fulyqualified domain name (FQDN). While the FQDN is preferred,many end systems do not have registered domain names, so IPaddresses are permitted. While Via header field tells other elements where to send response, the Contact header field tellsother elements where the called party can be reached directly.In a response, Via, To, From, Call-ID, and CSeq header fields arecopied from the INVITE request.In addition to DNS and location service lookups, proxy servers canmake flexible ³ routing decisions´ to decide where to send a request.For example, if Bob¶s SIP phone returned 486 (busy) response, thebiloxi.com proxy server could proxy the INVITE to Bob¶s voicemailserver. A proxy server can also send an INVITE to a number of locations at the same time. This type of parallel search is known asforking.

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Internet Telephony

After learning the end point addresses, the endpoints can communicate directly

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Internet Telephony

O verview of SIP Messaging Syntax

Text-basedo Similar to H TTP

SIP messageso message = start-line

*message-header CRLF[message-body]o start-line = request-line | status-line

Request-line specifies the type of requestThe response line

o The success or failure of a given request

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Internet Telephony

Message headerso Additional information of the request or responseo E.g.,

The originator and recipientRetry-after header Subject header

Message bodyo Describe the type of session

The media formatSDP, Session Description Protocol

o Could include an ISDN User Part messageo Examined only at the two ends

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Internet Telephony

SIP Requests

method SP request-URI SP SIP-version CRLFrequest-URI

o The address of the destinationMethods

o INVITE, ACK, O PTI O NS, BYE, CANCLE,REGISTERextensions: INF O , REFER, UPDATE, «

o INVITEInitiate a session

Information of the calling and called partiesThe type of mediaIAM (initial address message) of ISUP

ACK only the final response

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BYEo Terminate a sessiono Can be issued by either the calling or called party

O ptionso Query a server as to its capabilities

A particular type of mediaThe response if sent an INVITECANCEL

o Terminate a pending requesto E.g., an INVITE did not receive a final response

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REGISTERo Log in and register the address with a SIP server o ³ all SIP servers´ ± multicast address (224.0.1.1750)o Can register with multiple serverso Can have several registrations with one server

INFO

o RFC 2976o Transfer information during an ongoing session

DTMF digitsaccount balance informationmidcall signaling information generated in another network

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Internet Telephony

³O ne number´ service

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Internet Telephony

SIP Addressing

SIP URLs (Uniform Resource Locators)o user@hosto E.g.,

sip:[email protected]:[email protected]

o Supplement the URLsip:[email protected];user=phone

o sip:user:password@host:port;uri-parameters?headers

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Internet Telephony

Message H eaders

Provide further information about the messageo information elements

E.g.,o To:header in an INVITE

The called partyo From:header

The caling party

Four main categorieso General, request, response, and entity headerso

A list in Table 5-2o Mapping in Table 5-3

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Internet Telephony

General H eaders

Used in both requests and responsesBasic information

o E.g., To:, From:, Call-ID:, «Contact:

o A URL for future communicationo May be different from the From: header Requests passed through proxies

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Request H eaderso Apply only to SIP requestso Addition information about the request or the cliento E.g.,

Subject:Priority:, urgency of the request

Authorization:, authentication of the request originator

Response H eaderso Further information about the responseo E.g.,

Unsupported:, featuresRetry-After

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Internet Telephony

Entity H eader o Session information presented to the user o Session description, SDP

The RTP payload type, an address and porto Content-Length, the length of the message bodyo Content-Type, the media type of the messageo Content-Encoding, for message compressiono Content Disposition,o Content-Language,o

Allow, used in a Request to indicate the set of methods supportedo Expires, the date and time

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Internet Telephony

Example of SIP Message Sequences

Registrationo Via:o Call-ID:

host-specifico Content-Length:

Z ero, no msg bodyo Cseg:

Avoid ambiguityo Expires:

TTL0, unreg

o Contact:*

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Internet Telephony

Invitation

A two-party callo Subject:

optionalo Content-Type:

application/sdp

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Internet Telephony

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Termination of a Call

Cseq:o H as changed

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Redirect Servers

An alternative addresso 302, Moved temporarily

Another INVITEo Same Call-IDo Cseq ++

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Internet Telephony

Proxy Servers

Entity headers are omittedChanges the Req-URIVia:

o The patho Loop detected, 482o For a responseThe 1 st Via: header

Checkedremoved

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Internet Telephony

Proxy state

Can be either stateless or statefulRecord-Route:

o The messages and responses may not pass through the sameproxy

Use Contact:o A Proxy might require that it remains in the signaling path

In particular, for a stateful proxyo Insert its address into the Record-Route: header o The response includes the Record-Route: header o The Record-Route: header is used in the subsequent requestso The Route: header = the Record-Route: header in reverse

order, excluding the first proxyo Each proxy remove the next from the Route: header

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Internet Telephony

Forking Proxy

³ fork´ requests A user is registered atseveral locations

o ;branch=xxx

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The Session Description Protocol

The message bodyo SDP, RFC 2327

The Structure of SDPo Session Level Info

NameThe originator The time

o Media Level InfoMedia typePort number

Transport protocolMedia format

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Internet Telephony

SDP session description structure

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SDP Syntax

A number of lines of textIn each line

o field=valueSession-level fields first

Media-level fieldso Begin with media description field (m=)

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Internet Telephony

Mandatory Fields

v=(protocol version)o=(session origin or creator and session id)s=(session name), a text stringt=(time of the session)

o t=<start time> <stop time>o NTP time values in seconds

m=(media)o m=<media> <port> <transport> <fmt list>o Media typeo The transport porto The transport protocolo The media format, an RTP payload format

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Internet Telephony

O ptional Fileds

i=(session information)o A text descriptiono At both session and media levels

u=(URI of description)o Where further session information can be obtainedo O

nly at session levele=(e-mail address)o Who is responsible for the sessiono O nly at the session level

p=(phone number)o O nly at the session level

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Internet Telephony

c=(connection information)o Connection type, network type, and connection addresso At session or media level

b=(bandwidth information)o In kilobits per secondo At session or media level

r=<repeat interval> <active duration> <list of offsetsfrom start- time>

o For regularly scheduled sessiono H ow often and how many times

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z=(timezone adjustments)o z=<adjustment time> <offset> <adjustment time> <offset> ....o For regularly scheduled sessiono Standard time and Daylight Savings Time

k=(encryption key)o k=<method>:<encryption key>o An encryption key or a mechanism to obtain ito At session or media level

a=(attributes)o Describe additional attributes

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O rdering of Fields

Session Levelo Protocol version (v)o O rigin (o)o Session name (s)o Session information (i)o URI (u)o E-mail address (e)o Phone number (p)o Connection info (c)o Bandwidth info (b)o Time description (t)o Repeat info (r)o Time zone adjustments (z)o Encryption key (k)o Attributes (a)

Media levelo Media description (m)o Media info (i)o Connection info (c)

O ptional if specified at thesession level

o Bandwidth info (b)o Encryption key (k)o Attributes (a)

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Internet Telephony

Subfields

Field = <value of subfield1> <value of subfield2><value of subfield3> «

O rigin (o)o Username, the originator¶s login id or ³ -´o session ID

A unique IDMake use of NTP timestampo version, a version number for this particular sessiono network type

A text string; IN refers to Interneto address type

IP4, IP6o Address, a fully-qualified domain name or the IP addresso o=mhandley 2890844526 2890842807 IN IP4 126.16.64.4

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Internet Telephony

Connection Datao The network and address at which media data are to be

receivedo Network type, address type, connection addresso c=IN IP4 224.2.17.12/127

Media Informationo Media type

Audio, video, application, data, or controlo Port, 1024-65535o Format

List the various types of mediaRTP/AVP payload types

o m= audio 45678 RTP/AVP 15 3 0G.728, GSM, G.711

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Attributeso Property attribute

a=sendonlya=recvonly

o value attributea=orient:landscape

o rtpmap attributeThe use of dynamic payload typea=rtpmap:<payload type> <encoding name>/<clock rate>[/<encoding parameters>].m=video 54678 RTP/AVP 98a=rtpmap 98 L16/16000/2

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Usage of SDP with SIP

SIP for the establishment of multimediasessionsSDP ± a structured language for describing thesessions

o The entity header

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Negotiation of Media

Fig 5-15o G.728 is selected

If a mismatcho 488 or 606o Not Acceptableo A Warning header INVITE with multiple

media streamso Unsupported should also

be returnedo With a port number of zero

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O ffer/answer

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O PTI O NS Methodo Determine the capabilitie s

of a potential called party