Download It

Session Initiation Protocol (SIP)

Chapter 5

2Internet Telephony

Introduction

A powerful alternative to H.323 More flexible, simpler Easier to implement

Advanced features Better suited to the support of intelligent

user devices A part of IETF multimedia data and

control architecture SDP, RTSP (Real-Time Streaming

Protocol), SAP (Session Announcement Protocol)

3Internet Telephony

The Popularity of SIP

Originally Developed in the MMUSIC A separate SIP working group RFC 2543 Many developers

SIP + MGCP/MEGACO The VoIP signaling in the future

“back-off” Test products against each other Will be hosted by ETSI

4Internet Telephony

SIP Architecture

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

multimedia sessions SIP + SDP

Describe the session characteristics Separate signaling and media streams

5Internet Telephony

SIP Network Entities

Clients User agent clients Application programs sending SIP requests

Servers Responds to clients’ requests

Clients and servers may be in the same platform

Proxy Acts as both clients and servers

6Internet Telephony

Four types of servers Proxy servers

Handle requests or forward requests to other servers

Can be used for call forwarding

7Internet Telephony

Redirect servers Map the destination address to zero or more new

addresses Do not initiate any SIP requests

8Internet Telephony

A user agent server Accept SIP requests and contacts the user The user responds → an SIP response A SIP device E.g., an SIP-enabled telephone

A registrar Accepts SIP REGISTER requests

Indicating the user is at a particular address Typically combined with a proxy or redirect server

9Internet Telephony

SIP Call Establishment

It is simple A number of interim responses

10Internet Telephony

SIP Advantages

Attempt to keep the signaling as simple as possible

Various pieces of information can be included within the messages

Including non-standard information Enable the users to make intelligent decisions

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


Call Completion to Busy Subscriber service


Overview of SIP Messaging Syntax

Text-based Similar to HTTP

SIP messages message = start-line

*message-header CRLF

[message-body] start-line = request-line | status-line

Request-line specifies the type of request The response line

The success or failure of a given request


Message headers Additional information of the request or

response E.g.,

The originator and recipient Retry-after header Subject header

Message body Describe the type of session

The media format SDP, Session Description Protocol

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


SIP Requests

method SP request-URI SP SIP-version CRLF request-URI

The address of the destination Methods

INVITE, ACK, OPTIONS, BYE, CANCLE, REGISTER INVITE

Initiate a session Information of the calling and called parties The type of media ～ IAM (initial address message) of ISUP ACK only the final response


BYE Terminate a session Can be issued by either the calling or called party

Options Query a server as to its capabilities

A particular type of media The response if sent an INVITE

CANCEL Terminate a pending request E.g., an INVITE did not receive a final response

REGISTER Log in and register the address with a SIP server “all SIP servers” – multicast address (224.0.1.1750) Can register with multiple servers Can have several registrations with one server


SIP Responses

SIP version SP status code SP reason-phrase CRLF reason-phrase

A textual description of the outcome Could be presented to the user

status code, RFC 2543 A three-digit number 1XX Informational 2XX Success (only code 200 is defined) 3XX Redirection 4XX Request Failure 5XX Server Failure 6XX Global Failure All responses, except for 1XX, are considered final

Should be ACKed


“One number” service


SIP Addressing

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

sip:[email protected] sip:[email protected]

Supplement the URL sip:[email protected];user=phone


Message Headers

Provide further information about the message ～ information elements

E.g., To:header in an INVITE

The called party From:header

The caling party

Four main categories General, request, response, and entity headers A list in Table 5-2 Mapping in Table 5-3


General Headers

Used in both requests and responses Basic information

E.g., To:, From:, Call-ID:, … Contact:

A URL for future communication May be different from the From: header

Requests passed through proxies


Request Headers Apply only to SIP requests Addition information about the request or the

client E.g.,

Subject: Priority:, urgency of the request Authorization:, authentication of the request originator

Response Headers Further information about the response E.g.,

Unsupported:, features Retry-After


Entity Header Session information presented to the user Session description, SDP

The RTP payload type, an address and port Content-Length, the length of the message

body Content-Type, the media type of the message Content-Encoding, for message compression Content Disposition, Content-Language, Allow, used in a Request to indicate the set

of methods supported Expires, the date and time


Example of SIP Message Sequences

Registration Via: Call-ID:

host-specific Content-Length:

Zero, no msg body Cseg:

Avoid ambiguity Expires:

TTL 0, unreg

Contact: *


Invitation

A two-party call Subject:

optional Content-Type:

application/sdp


Termination of a Call

Cseq: Has changed


Redirect Servers

An alternative address

302, Moved temporarily

Another INVITE Same Call-ID Cseq ++


Proxy Servers

Entity headers are omitted

Changes the Req-URI Via:

The path Loop detected, 482 For a response

The 1st Via: header Checked removed


Proxy state

Can be either stateless or stateful Record-Route:

The messages and responses may not pass through the same proxy

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

path In particular, for a stateful proxy

Insert its address into the Record-Route: header The response includes the Record-Route: header The Record-Route: header is used in the subsequent

requests The Route: header = the Record-Route: header in

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


Forking Proxy

“fork” requests A user is registered at

several locations ;branch=xxx


The Session Description Protocol

The message body SDP, RFC 2327

The Structure of SDP Session Level Info

Name The originator The time

Media Level Info Media type Port number Transport protocol Media format


SDP Syntax

A number of lines of text In each line

field=value Session-level fields first Media-level fields

Begin with media description field (m=)


Mandatory Fields

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

stop time m=(media)

Media type The transport port The transport protocol The media format, an RTP payload format


Optional Fileds

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

u=(URI of description) Where further session information can be obtained Only at session level

e=(e-mail address) Who is responsible for the session Only at the session level

p=(phone number) Only at the session level


c=(connection information) Connection type, network type, and connection

address At session or media level

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

r=(repeat times) For regularly scheduled session How often and how many times

z=(timezone adjustments) For regularly scheduled seesion Standard time and Daylight Savings Time


k=(encryption key) An encryption key or a mechanism to obtain it At session or media level

a=(attributes) Describe additional attributes


Ordering of Fields

Session Level Protocol version (v) Origin (o) Session name (s) Session information (i) URI (u) E-mail address (e) Phone number (p) Connection info (c) Bandwidth info (b) Time description (t) Repeat info (r) Time zone adjustments

(z) Encryption key (k) Attributes (a)

Media level Media description (m) Media info (i) Connection info (c)

Optional if specified at the session level

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


Subfields

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

Origin (o) Username, the originator’s login id or “-” session ID

A unique ID Make use of NTP timestamp

version, a version number for this particular session network type

A text string IN refers to Internet

address type IP4, IP6

Address, a fully-qualified domain name or the IP address


Connection Data The network and address at which media data are

to be received Network type Address type Connection address

Media Information Media type

Audio, video, application, data, or control Port, 1024-65535 Format

List the various types of media RTP/AVP payload types

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


Attributes Property attribute

a=sendonly a=recvonly

value attribute a=orient:landscape

rtpmap attribute The use of dynamic payload type a=rtpmap:<payload type> <encoding name>/<clock

rate> [/<encoding parameters>]. m=video 54678 RTP/AVP 98 a=rtpmap 98 L16/16000/2


Usage of SDP with SIP

SIP for the establishment of multimedia sessions

SDP – a structured language for describing the sessions The entity header


Negotiation of Media

Fig 5-15 G.728 is selected

If a mismatch 488 or 606 Not Acceptable A Warning header

INVITE with multiple media streams

Unsupported should also be returned

With a port number of zero


Options Method Determine the

capabilities of a potential called party


Usage of SIP for Features/Services

Personal mobility by registration Can carry MIME (Multi-Purpose Internet

Mail Extension) content Text, HTML documents, an image, etc.

SIP address is a URL Click-to-call applications

Supplementary Custom Local Area Signaling Service (CLASS) services Call waiting, call forwarding, multi-party

calling, call screening Proxy-controlled: QoS, IN SCP, INAP


Call Forwarding

On busy 486, busy here


Consultation Hold

C=0 An address


SIP Extensions and Enhancements

RFC 2543, March 1999 Proposed standard, March 2002 SIP-T

Include various extensions Will be enhanced considerably before it

becomes an Internet standard


183 Session-Progress Message

The addition of a new response Status code 183 To open a one-way media path

From the called party to calling party Enable in-band call progress information to be

transmitted Tones or announcements

ACM (address complete message) of SS7 For SIP – PSTN – SIP connections

When a temporary media stream is needed Note that alerting signal can be

Status code 180 (ringing) The temporary media stream will be terminated

As soon as the called user answers


SIP INFO Method

A new SIP method The transfer of information in the middle of a

call DTMF digits, account-balance information,

mid-call signaling information (from PSTN) A powerful, flexible tool to support new

services


The SIP Supported Header

The Require header a client indicates to a server that the server

must support certain features In responses 421, extension required

The Supported header For server to know a client’s capabilities Included in both requests and responses

BYE, CANCEL, INVITE, OPTIONS and REGISTER Should not be included in the ACK


Reliability of Provisional Responses

Provisional Responses 100 (trying), 180 (ringing), 183 (session in

progress) Are not answered with an ACK

If the messages is sent over UDP Unreliable

Lost provisional response may cause problems when interoperating with other network 180, 183 → Q931 alerting or ISUP ACM To drive a state machine E.g., a call to an unassigned number

ACM to create a one-way path


RSeq Response seq +1, when retxm

Rack Response ACK

PRACK Prov Resp ACK

Should not Apply to 100


Integration of SIP an Resource Mang

The signaling might take a different path from the media

Assume resource-reservation mechanisms available (Chapter 8)

A new SIP header in the INVITE Resources reservation is needed The user should not yet be alerted But unrecognized header is ignored


Integration of Resource Management and SIP for IP Telephony A new method, PRECONDITION-MET The far-end phone will not ring until Also specifies extensions to SDP Can define any number of preconditions in

SDP without revise SIP every time Being sent end-to-end “a=qos:” strength-tag SP direction-tag [SP

confirmation-tag] “a=secure:” strength-tag SP direction-tag [SP

confirmation-tag] If failed, could select a lower-bandwidth codec


Interworking

PSTN Interworking A SIP URL A network gateway Fig. 5-23

SIP to PSTN call Fig. 5-24

PSTN to SIP call

PSTN – SIP – PSTN MIME media types For ISUP and QSIG


Interworking with H.323

An Internet draft SIP-H.323 interworking gateway


Summary

The future for signaling in VoIP networks Simple, yet flexible Easier to implement Fit well with the media gateway control

protocols


Reference Architecture of R00

GfGi

Iu

GiMr

Gi

Ms

Gi

R UuMGW

Gn

Gc

Signalling and Data TransferInterface

SignallingInterface

TE MT UTRAN

Gr

SGSN GGSN

EIR

MGCF

R-SGW *)

MRF

MultimediaIP Networks

PSTN/Legacy/External

Applications &Services *)

Mm

Mw

Legacy mobilesignallingNetw ork

Mc

Cx

AlternativeAccess

Network

Mh

CSCF

CSCF

Mg

T-SGW *)

T-SGW *)

HSS *)

HSS *)

Applications& Services *)

MSC server GMSC server

McMc

D C

SCP

CAP

MGWNb

Nc

Iu

Iu

R-SGW *)Mh

CAPCAP

R UmTE MT

BSS/GERAN

Gb

A

*) those elements are duplicated for figurelayout purpose only, they belong to the samelogical element in the reference model

Iu


Support of Roaming Subscribers

via the service platform in the Home Network

via an external service platform

UE

P-CSCF

ServingCSCF

HomeNetwork

VisitedNetwork

ExternalServicePlatform

Gm

Mw


UE Accessing IM Subsystem Services

In the visited network

Home NetworkIM Subsystem

Visited NetworkIM Subsystem

Inter-NetworkIM Backbone

Internet

Intranets

UE

GGSN

BG

BG

SGSN

PDP Context

Visited Network

Gi

Virtual presence of UEin visited network IM subsystem(UE’s IP-address is here)


Public and Private Identities

One private user identity IMSI within a NAI

One or more public user identities

DOCUMENTTYPE

TypeUnitOrDepartmentHereTypeYourNameHere TypeDateHere

Private user identityIM subscription

Public user identity 1 (e.g. SIP URL)

Public user identity 2 (e.g. E.164)

Public user identity 3. . .


Roles of CSCF

Call State Control Function SIP proxy servers

Proxy CSCF The first contact point within the IM

subsystem Forward the SIP register requests from the UE

to an I-CSCF Forward the SIP messages from the UE to the

S-CSCF CDR FFS

Authorization, QoS management, security


Interrogating CSCF The contact point within an operator’s

network Assign an S-CSCF to a user performing SIP

Reg Obtain from HSS the address of the S-CSCF Forward SIP messages to the S-CSCF Forward SIP messages to the MGCF CDR FFS

Inter-operator security


Serving CSCF Perform the session control services A registrar Interaction with Services Platforms On behalf of an originating endpoint On behalf of an destination endpoint CDR FFS

Security issues


Registration Information Flow

P-CSCF HSS

1. REGISTER

UE

Visited Network Home Network

7. Continuation of registration

I-CSCF

5. Cx-Select-Pull

6. Cx-Select-Pull- Resp

3. Cx-Query

4. Cx-Query-Resp

2 REGISTER


Registration

P-CSCF HSSI-CSCF

5. Cx-Pull

6. Cx-Pull Resp

2. Register

S-CSCFUE

Visited Network Home Network

7. 200 OK

8. 200 OK9. 200 OK

1. S-CSCF selection

3. Cx-put

4. Cx-put Resp


Call flow

3. INVITE

21. Ringing

S-CSCF#1 S-CSCF#2

10. SDP

16. Success

27. 200 OK

13. Final SDP

31. ACK

I-CSCF HSS

4. Location Query5. Response

6. INVITE

11. SDP

1. INVITE

12. SDP

8. INVITE

9. SDP

19. Ringing

24. Ringing

25. 200 OK

30. 200 OK

Home NetworkOriginating Network Terminating Network

22. Ringing

28. 200 OK

2. Service Control

7. Service Control

20. Service Control

23. Service Control

26. Service Control

29. Service Control

17. Resource Reservation Successful18. Success

14. Final SDP15. Final SDP

32. ACK33. ACK


3. INVITE

21. Ringing

S-CSCF#1 S-CSCF#2

10. SDP

16. Success

27. 200 OK

13. Final SDP

31. ACK

I-CSCF HSS

4. Location Query5. Response

6. INVITE

11. SDP

1. INVITE

12. SDP

8. INVITE

9. SDP

19. Ringing

24. Ringing

25. 200 OK

30. 200 OK

Home NetworkOriginating Network Terminating Network

22. Ringing

28. 200 OK

2. Service Control

7. Service Control

20. Service Control

23. Service Control

26. Service Control

29. Service Control

17. Resource Reservation Successful18. Success

14. Final SDP15. Final SDP

32. ACK33. ACK


Different Kinds of CSCFs

P-SCSFP-SCSFVisited B

S-CSCFS-CSCF

Home A

1

2

7

10

11 12

13

15

17

P-CSCFP-CSCFVisited A

18

AB

GGSNGGSNSGSNSGSN

Radio Access NetworkRadio Access Network

GGSNGGSNSGSNSGSN

Radio Access NetworkRadio Access Network

S-CSCFS-CSCF

Home B

8

I-CSCFI-CSCF

HSSHSS

9

14

6

34

I-CSCFI-CSCF

HSSHSS

5

16


Interwork with PSTN

1. IAM

11. IP-ACM

2. IP-IAM

PSTN T-SGW MGW

6. Final SDP

10. Ringing

14. IP-ANM15. ANM

9. Success

17. ACK

MGCF

4. INVITE

5. SDP

13. 200 OK

12. ACM

Home Network

16. H.248 interaction tomodify connectionto start media flow

3. H.248 interaction tocreate connection

7. H.248 interaction tomodify connection

to reserve resources

8. ResourceReservation


1. IAM

11. IP-ACM

2. IP-IAM

PSTN T-SGW MGW

6. Final SDP

10. Ringing

14. IP-ANM15. ANM

9. Success

17. ACK

MGCF

4. INVITE

5. SDP

13. 200 OK

12. ACM

Home Network

16. H.248 interaction tomodify connectionto start media flow

3. H.248 interaction tocreate connection

7. H.248 interaction tomodify connection

to reserve resources

8. ResourceReservation

Download It

Documents

Transcript of Download It