Sip（session intial protocol）

7/24/2019 Sip session intial protocol

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Session Initiation Protocol


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Goal We want to know

SIP Architecture SIP Messages Syntax

Basic Call Example Session Description Protocol (SDP)

SIP Extensions


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TCP

IPv4/IPv6 support mobility

UDP

Q

oS

RTSPH.323......Web

Applications

RTCP RTPSDP

SIP Megaco

Wireless Access

3GPP/3GPP2iMAX W i Fi

Layer 2

Layer 1

Media

Audio, Video

RSVP

ADSL Cable

Residential Access

Ethernet

Company Access

SignalingQuality of

ServiceMedia

Transport

Multimedia protocol stack


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SIP Architecture

Introduction to SIP

SIP Network Entities

Simple Example


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Introduction

A powerful alternative to H.323 More flexible, simpler

Easier to implement Advanced features

Better suited to the support of intelligent userdevices

A part of IETF multimedia data and controlarchitecture

SDP (Session Description Protocol), RTSP (Real-Time Streaming Protocol), SAP (SessionAnnouncement Protocol)


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The Popularity of SIP

Originally Developed in the MMUSIC (MultipartyMultimedia Session Control) A separate SIP working group

RFC 2543 The latest version: RFC 3261

SIP + MGCP/MEGACO The VoIP signaling in the future

bake-off or SIPit (SIP Interoperability Tests) Various vendors come together and test their products

against each other To ensure that they have implemented the specification

correctly

To ensure compatibility with other implementations


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SIP Basic (1/2)

SIP is a peer-to-peer protocol End-devices initiate sessions SIP is an application layersignaling protocol

Create, modify and terminate sessions

Applications can be voice, video, gaming, instantmessaging, presence, call control, etc.

SIP uses existing IETF protocols to provide: Message formatting (HTTP 1.1)

Name resolution and mobility (DHCP and DNS) Media (RTP) Media negotiation (Session Description Protocol

SDP) Application encoding (MIME)


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SIP Basic (2/2)

SIP is ASCII text-based Eases implementation and debugging Uses URI style addresses and syntax Flexible transport

can use UDP, TCP, TLS, or SCTP Uses SDP for describing media sessions

Audio, video, realtime text ...

Simple extensible protocol

MethodsDefine transaction HeadersDescribe transaction

BodySDP and other MIME content


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SIP Architecture

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

sessions

SIP + SDP

Describe the session characteristics Separate signaling and (RTP) media streams

SIP Signaling

IP Network

RTP Media Stream

SIP User SIP User


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Client-Server Model Clients

User agent clients (UAC)

Application programs sending SIP requests Servers

Responds to clients requests

Four types of servers


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Types of Servers (1/3)

Proxy serversProxy servers Handle requests or forward requests to other servers

Can be used for call forwarding, time-of-day routing, orfollow-me services

[email protected] [email protected]

1.Request [email protected]


4.Response 3.Response


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Redirect serversRedirect servers Map the destination address to zero or more new

addresses

Redirect Server


[email protected]

[email protected]

2.Moved temporarily Contact: [email protected]

3.ACK

4.Request

[email protected]

5.Response


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A user agent server (UAS)A user agent server (UAS) Accepts SIP requests and contacts the user

The user responds an SIP response

Usually, a SIP device = UAC+UAS

A registrarA registrar Accepts SIP REGISTER requests

Indicating that the user is at a particular address

Personal mobility

Typically combined with a proxy or redirect server


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SIP Call Establishment

It is simple, which contains a number of interimresponses.

SIPdevice A

SIPdevice B

a

b

c

d

e

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INVITE

Ringing

OK

ACK

Conversation

BYE

OK


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Call Completion to Busy SubscriberService

a

b

c

d

e

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g

Conversation

BYE

OK

ACK

OK

Ringing

INVITE

Busy (Try at 4pm)

INVITE

ACK

j

i

h

SIP

device A

SIP

device B


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SIP Advantages

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

Does not care what type of media is to be exchangedduring a session or the type of transport to be used for the

media Various pieces of information can be included within

the messages Including non-standard information

Enable the users to make intelligent decisions

The control of the intelligent features is placed in the handsof the customer, not the network operator.

E.g., SUBJECT header


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Overview of SIP Messaging Syntax

SIP Message Syntax

SIP Request Messages

Simple Example


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SIP Messaging Syntax (1/2)

Text-based Similar to HTTP (Hypertext Transfer Protocol)

Disadvantage more bandwidth consumption

SIP messages message = start-line

*message-header CRLF

[message-body]

start-line = request-line | status-line Request-line specifies the type of request

Response line (status-line) indicates the success orfailure of a given request.


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SIP Messaging Syntax (2/2)

Message headers Additional information of the request or response

E.g.,

From: /To: headers for the originator and recipient

Retry-after header Subject header

Message body Describe the type of session

The most common structure for the message body is SDP(Session Description Protocol).

Could include an ISDN User Part message

Examined only at the two ends


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SIP Requests

Method SP Request-URI SP SIP-version CRLF

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

Request-URI

The address of the destination

Methods RFC 2543: INVITE,ACK, OPTIONS, BYE, CANCLE, REGISTER

RFC 2976: INFO

RFC 3261: SUPPORT,

RFC 3262: PRACK RFC 3265: SUSCRIBE & NOTIFY

RFC 3311: UPDATE

RFC 3428: MESSAGE

RFC 3515: REFER


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SIP Methods (1/2)

INVITEINVITE Initiate a session

Information of the calling and called parties

The type of media

Similar to IAM (initial address message) of ISUP

ACKACK Only when receiving the final response

BYEBYE Terminate a session

Can be issued by either the calling or called party


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SIP Methods (2/2)

OPTIONSOPTIONS Query a server as to its capabilities

A particular type of media

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

REGISTERREGISTER

Log in and register the address with a SIP server all SIP servers multicast address (224.0.1.175)

Can register with multiple servers

Can have several registrations with one server


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One number ServiceRegistrar/Proxy CallerUser at Address 1User at Address 2

OK

Register (address 1)

Register (address 2)

OK

INVITE

Trying

INVITE

INVITE

OK

CANCEL

OK (for INVITE)

OK (for CANCEL)

ACK

ACK

Conversation


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SIP Responses

SIP-Version SP Status-Code SP Reason-PhraseCRLF

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

Response: SIP/2.0 200 ok

Status-Code A three-digit number

Reason-Phrase A textual description of the outcome

Could be presented to the user


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Status Code

Provisional response 1XX Informational (ex: 181 Ringing)

Final response 2XX Success (ex: 200 OK)

3XX Redirection (ex: 302 Moved temporarily)

4XX Client Error (ex: 401 Unauthorized)

5XX Server Error (ex: 505 SIP version not supported)

6XX Global Failure (ex: 604 Does not exist anywhere)

All responses, except for 1XX, are considered finaland Should be ACKed.


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SIP Addressing

SIP URIs (Uniform Resource Identifier)

user@host

Also called as URLs (Uniform Resource Locators)

Two types of SIP URIs:

Address of Record (AOR) (identifies a user) sip:[email protected]

Contact (identifies a device and is usually a Fully QualifiedDomain Name, FQDN) sip:[email protected] or sip:[email protected]

Other example:

sip:[email protected]

sip:[email protected];user=phone

sips:[email protected]


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Message HeadersMessage Headers

Provide further information about the message Ex: Subject: vacation

Ex: Content-Type:application/sdp

Four main categories General, Request, Response, and Entity headers


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General Headers

Used in both requests and responses

To, From, CSeq, Call-ID, and Via are mandatory for all SIPmessages.

Basic information

To: headerin a REGISTER indicates the address-of-record ofthe user.

Ex: To: [email protected]

To: headerin an INVITE indicates the called party.

Ex: To: Boss

Ex: To: Boss

From: headerindicates the originator.

Ex: From: Daniel;tag=4455


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To: Header

The To headerfield first and foremost specifies

The desired "logical" recipient of the request

The address-of-record of the user or resource that is the target ofthis request.

The original recipient may or may not be the UAS processingthe request, due to call forwarding or other proxy operations.

Request-URI identifies the UAS that is to process the request.


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Request-URI

Request-URI identifies the UAS that is to processthe request.

The initial Request-URI of the message SHOULDbe set to the same value of the URI in the To field.

One notable exception is the REGISTER

A UA uses the REGISTER method to bind itsaddress-of-record to a specific contact address.


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Contact: Header

Provides a URI for use in future communication regarding aparticularsession

Ex1: In a SIP INVITE, the Contact header might be differentfrom the From header.

An third-party administrator initiates a multiparty session.

Ex 2: Used in response, it is useful for directing furtherrequests directly to the called user.

Ex 3: It is used to indicate a more appropriate address if anINVITE issued to a given URI failed to reach the user.


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Request / Response Headers

Request Headers

Apply only to SIP requests

Addition information about the request or the client

Ex: Subject: vacation

Ex: Priority: (emergency, urgent, normal, or non-urgent)

Response Headers

Further information about the response that cannot be included

in the status line Ex: Unsupported

Ex: Retry-After


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Entity Headers

Indicate the type and format of information includedin the message body

Content-Length: the length of the message body

Content-Type: media type of the message body Ex: Content-Type:application/sdp

Content-Encoding: for message compression

Content Disposition: how a message part should beinterpreted

session, icon, alert, render


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Examples

Registration

Invitation

Termination of a Call


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[email protected]

REGISTER sip:registrar.work.com SIP/2.0Via: SIP/2.0/UDP station1.work.com; branch=z9hG4bK123

Max-Forwards: 70From: sip:[email protected]; tag = 123456To: sip:[email protected]: 123456@ station1.work.comCSeq: 1 REGISTERContact: sip:[email protected]: 7200Content-Length: 0

SIP/2.0 200 OKVia: SIP/2.0/UDP station1.work.com; branch=z9hG4bK123From: sip:[email protected]; tag = 123456To: sip:[email protected]: 123456@ station1.work.comCSeq: 1 REGISTERContact: sip:[email protected]: 3600Content-Length: 0

a

b


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Invitation

A two-party call

Subject:

optional

Content-Type:

application/sdp

Transaction: Command Sequence

A dialog ID

To identify a peer-to-peer

relationship between twouser agents

Tag in From

Tag in To

Call-ID

BossDaniel

INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP station1.work.com; branch=z9hG4bK123Max-Forwards: 70From: Daniel; tag=44551Contact: sip:[email protected]: BossCall-ID: [email protected]: 1 INVITESubject: VacationContent-Length: xxxContent-Type: application/sdp

Content-Disposition: session(message body)

a

bSIP/2.0 180 RingingVia:SIP/2.0/UDP station1.work.com;branch=z9hG4bK123From: Daniel; tag=44551To: Boss; tag=11222Contact: sip:[email protected]: [email protected]: 1 INVITEContent-Length: 0

cSIP/2.0 200 OKVia: SIP/2.0/UDP station1.work.com; branch=z9hG4bK123From: Daniel; tag=44551To: Boss; tag=11222Contact: sip:[email protected]: [email protected]: 1 INVITESubject: VacationContent-Length: xxxContent-Type: application/sdpContent-Disposition: session(message body)

dACK sip:[email protected] SIP/2.0Via:SIP/2.0/UDP station1.work.com;branch=z9hG4bK123Max-Forwards: 70From: Daniel; tag=44551To: Boss; tag=11222Call-ID: [email protected]: 1 ACKContent-Length: 0

Conversation


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Termination of a CallBossDaniel

BYE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP station1.work.com; branch=z9hG4bK123Max-Forwards: 70From: Daniel; tag=44551To: Boss; tag=11222Call-ID: [email protected]: 2 BYEContent-Length: 0

a

bSIP/2.0 200 OK

Via: SIP/2.0/UDP station1.work.com; branch=z9hG4bK123From: Daniel; tag=44551To: Boss; tag=11222Call-ID: [email protected]: 2 BYEContent-Length: 0


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Redirect and Proxy Servers

Redirect serverProxy server


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Redirect server

An alternative address 302 Moved temporarily

Another INVITE

Same Call-ID CSeq ++

INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP station1.work.com; branch=z9hG4bK124Max-Forwards: 70From: Daniel; tag=44551Contact: sip:[email protected]: BossCall-ID: [email protected]: 2 INVITESubject: Vacation

Content-Length: xxxContent-Type: application/sdpContent-Disposition: session(message body)

c

d

ACK sip:[email protected] SIP/2.0Via: SIP/2.0/UDP station1.work.com; branch=z9hG4bK123Max-Forwards: 70From: Daniel; tag=44551To: BossCall-ID: [email protected]: 1 ACK

Boss

Daniel

INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP station1.work.com; branch=z9hG4bK123Max-Forwards: 70From: Daniel; tag=44551Contact: sip:[email protected]

To: BossCall-ID: [email protected]: 1 INVITESubject: VacationContent-Length: xxxContent-Type: application/sdpContent-Disposition: session(message body)

a

bSIP/2.0 302 Moved Temporarily

Via:SIP/2.0/UDP station1.work.com;branch=z9hG4bK123From: Daniel; tag=44551To: Boss; tag=11222Call-ID: [email protected]: 1 INVITEContact: sip:[email protected]

Redirect Server


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Proxy server

Sits between a user-agent client and the far-enduser-agent server

Numerous proxies can reside in a chain betweenthe caller and callee.

The proxy may change the Request-URI. Via: header

The path taken by a request

Loop detected, 482 (status code) For a response

The 1st Via: header, Checked, Removed

Branch: used to distinguish between multiple responses to

the same request/ to detect loop

p4p3p2p1caller

INVITE

482

ACK


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Forking server

Forking Proxy: Issue a single request to multipledestinations A user is registered at several locations

fork requests

Branch: used to distinguish between multipleresponses to the same request Ex: on Via: header ;branch=z9hG4bK123

In order to handle such forking, a proxy must bestateful.


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Proxy state

Transaction stateless Proxy server forwards all methods and responses without

interaction

Transaction stateful Maintains state for the transaction until the final response

is received

Transaction Control Block (TCBTCB)


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Record-Route: and Route:

Record-Route:

Use Contact: the messages may not pass through thesame proxy

A stateful proxy might require that it remains in thesignaling path

Insert its address into the Record-Route: header

The response includes the Record-Route: header

The information contained in the Record-Route: header isused in the subsequent requests related to the same call.

The Route: header = the Record-Route: header inreverse order


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Through a Proxy (1/2)Boss Daniel

INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK7890Max-Forwards: 70From: Boss; tag=ab12Contact: BossTo: DanielCall-ID: [email protected]

CSeq: 1 INVITE

a

b

SIP/2.0 100 TryingVia: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK7890From: Boss; tag=ab12To: DanielCall-ID: [email protected]: 1 INVITE

c

sip:Server.work.com

INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP server.work.com; branch=z9hG4bKxyz1Via: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK7890

Max-Forwards: 69Record-route: From: Boss; tag=ab12Contact: BossTo: DanielCall-ID: [email protected]: 1 INVITE

SIP/2.0 200 OKVia: SIP/2.0/UDP server.work.com; branch=z9hG4bKxyz1Via: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK7890Record-route: From: Boss; tag=ab12To: Daniel; tag=xyz45Call-ID: [email protected]: 1 INVITEContact: sip:[email protected]

SIP/2.0 200 OK...

e

d


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Through a Proxy (2/2) Boss Daniel

ACK sip:[email protected] SIP/2.0Via: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK7891Max-Forwards: 70Route: From: Boss; tag=ab12To: Daniel; tag=xyz45Call-ID: [email protected]

CSeq: 1 ACK

e

f

sip:Server.work.com

ACK sip:[email protected] SIP/2.0Via: SIP/2.0/UDP server.work.com; branch=z9hG4bKxyz2Via: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK7891Max-Forwards: 69From: Boss; tag=ab12To: Daniel; tag=xyz45Call-ID: [email protected]: 1 ACK

SIP/2.0 200 OKVia: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK7890Record-route: From: Boss; tag=ab12To: Daniel; tag=xyz45Call-ID: [email protected]

CSeq: 1 INVITEContact: sip:[email protected]

g


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Multiple Locations (1/2)Boss

pc1

INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK789Max-Forwards: 70From: Boss; tag=ab12Contact: BossTo: DanielCall-ID: [email protected]: 1 INVITE

a

b

SIP/2.0 100 TryingVia: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK789From: Boss; tag=ab12To: DanielCall-ID: [email protected]: 1 INVITE

c

sip:Server.work.com

INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP server.work.com; branch=z9hG4bK123Via: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK789Max-Forwards: 69Record-coute: From: Boss; tag=ab12Contact: BossTo: Daniel

Call-ID: [email protected]: 1 INVITEd

INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP server.work.com; branch=z9hG4bK456Via: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK789Max-Forwards: 69Record-coute: From: Boss; tag=ab12Contact: BossTo: DanielCall-ID: [email protected]: 1 INVITE

pc2


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Multiple Locations (2/2) Boss

pc1

e

f

g

sip:Server.work.com

CANCEL sip:[email protected] SIP/2.0Via: SIP/2.0/UDP server.work.com; branch=z9hG4bK456Max-Forwards: 69Record-coute: From: Boss; tag=ab12

Contact: BossTo: DanielCall-ID: [email protected]: 1 CANCEL

SIP/2.0 200 OKVia: SIP/2.0/UDP server.work.com; branch=z9hG4bK456Via: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK789Record-coute: From: Boss; tag=ab12To: Daniel; tag=xyz45Call-ID: [email protected]: 1 INVITEContact: sip:[email protected]

pc2

SIP/2.0 200 OKVia: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK789Record-coute: From: Boss; tag=ab12To: Daniel; tag=xyz45Call-ID: [email protected]: 1 INVITEContact: sip:[email protected]


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Session Description Protocol (SDP)

Introduction to SIP


Simple Example


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The Session Description Protocol

The Most Common Message Body Be session information describing the media to be

exchanged between the parties

SDP, RFC 2327 (initial publication) (3266 4566)

SDP simply provides a format for describing sessioninformation to potential session participants.

RFC 3264 (An Offer/Answer Model with SDP) SIP uses SDP in an answer/offer mode.

An agreement between the two parties as to the types ofmedia they are willing to share


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The Structure of SDP

Text-based Protocol

The Structure of SDP Session Level Information

Name of the session

Originator of the session Time that the session is to be

active

Media Level Information

Media type Port number

Transport protocol

Media format

Originator and Session ID

Protocol Version

Session Name

Session Time

Media Name and Transport

Connection Information

Media Name and Transport

Connection Information

Session Description

Session Level Information

Media Description 1

Media Description 2


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SDP Syntax

A number of lines of text

In each line field=value

Ex: v=0

field= ...

Ex: m=audio 45678 RTP/AVP 0

field is exactly one character(case-significant)

Session-level fields Begin with version description field (v=)

Media-level fields

Begin with media description field (m=)


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Mandatory Fields (1/3)

v, o, s, t, m are mandatory fields.

v=(protocol version) (ex: v=0)

s=(session name) (ex: s= )

A text string for multicast conference

t=(time of the session) (ex: t=0 0)

The start time and stop time for pre-arranged multicast

conference


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Mandatory Fields (2/3) o=(session origin or creator)

Ex: o=collins 123456 001 IN IP4 s1.yy.com Username (ex: collins): the originators login id or -

Session ID (ex: 123456)

Make use of NTP timestamp as a unique ID

Version (ex: 001) Network type (ex: IN)

IN refers to Internet

Address type (ex: IP4)

IP4, IP6

address (ex: s1.yy.com)

A fully-qualified domain name or the IP address


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Mandatory Fields (3/3)

m=(media) Ex: m=audio 45678 RTP/AVP 15 3 0

Media type (ex: audio) Audio, video, application, data, control

The transport port (ex: 45678)

The transport protocol (ex: RTP/AVP)

The media format, an RTP payload format

(ex: 15 3 0)

List the various types of media format that can be supported


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Optional Fields (1/4)

Some optional fields can be applied at both session and

media levels. The value applied at the media level overrides that at the

session level for that media instance.

c=(connection data) (ex: c=IN IP4 s1.yy.com)

The network and address at which media data will be received

Network type (ex: IN)

Address type (ex: IP4)

Connection address (ex: s1.yy.com)

At session or media level


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a=(attributes) (ex: a=send only) Describe additional attributes

Property attribute

Ex: a=sendonly / a=recvonly

Value attribute

Used in a shared whiteboard session

Ex: a=orient:portrait / a=orient:landscape

rtpmap provides additional information for dynamic payloadtype.

Ex: m=video 54678 RTP/AVP 98

a=rtpmap:98 L16/16000/2

Dynamic RTP payload type=98

16-bit linear encoded stereo (2 channels) audio sampled at 16kHz


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i=(session information)

A text description At both session and media levels

It would be somewhat superfluous, since SIP already supports theSubject header.

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


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p=(phone number)

Only at the session level

b=(bandwidth information)

In kilobits per second


r=(repeat times)

For regularly scheduled session a session is to be repeated

How often and how many times


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z=(timezone adjustments)

For regularly scheduled session Standard time and daylight savings time

k=(encryption key)

An encryption key or a mechanism to obtain it for the purposesof encrypting and decrypting the media



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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 thesession level

Bandwidth info (b)

Encryption key (k)

Attributes (a)


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Usage of SDP with SIP

Simple Example

Negotiation of Media


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Usage of SDP with SIP

SIP and SDP make a wonderful partnership for the

transmission of session information. SIP provides the messaging mechanism for the establishment

of multimedia sessions.

SDP provides a structured language for describing thesessions.

The entity headers identifies the message body.

Ex: Content-Type:application/sdp


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Offer/Answer Model

Offer a selection of media formats

Answer which the receiver is willing to accept Supported media type: returned with a transport port

Unsupported media type: also be returned with a port

number of zero

(1) Offer

(2) Answer

Sender Receiver


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Daniel Boss

INVITE sip:[email protected] SIP/2.0From: Daniel; tag = abcd1234To: BossCSeq: 1 INVITEContent-Length: 213

Content-Type: application/sdpContent-Disposition: session

v=0o=collins 123456 001 IN IP4 station1.work.coms=c=IN IP4 station1.work.com

t=0 0m=audio 4444 RTP/AVP 2a=rtpmap 2 G726-32/8000m=audio 4666 RTP/AVP 4a=rtpmap 4 G723/8000m=audio 4888 RTP/AVP 15a=rtpmap 15 G728/8000

a

b

SIP/2.0 200 OK

SDP Inclusion in SIP Messages (1/2)


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SIP/2.0 200 OKFrom: Daniel; tag = abcd1234To: Boss; tag = xyz789CSeq: 1 INVITEContent-Length: 163Content-Type: application/sdp

Content-Disposition: session

v=0o=collins 45678 001 IN IP4 station2.work.coms=c=IN IP4 station2.work.comt=0 0

m=audio 0 RTP/AVP 2m=audio 0 RTP/AVP 4m=audio 6666 RTP/AVP 15a=rtpmap 15 G728/8000

b

c

d

ACK sip:[email protected] SIP/2.0From: Daniel; tag = abcd1234To: Boss; tag = xyz789CSeq: 1 ACK

Content-Length: 0

Conversation

Daniel Boss

SDP Inclusion in SIP Messages (2/2)

port=4888 port=6666


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Alternative Ways

Offerm=audio 4444 RTP/AVP 2 4 15a=rtpmap 2 G726-32/8000

a=rtpmap 4 G723/8000a=rtpmap 15 G728/8000

Answerm=audio 6666 RTP/AVP 15a=rtpmap 15 G728/8000


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Negotiation of Media (1/3)

Re-INVITE is issued when the server replies with

more than one codec. With the same dialog identifier(To and From tag), Call-ID

and Request-URI

Session version in o field is increased by 1.

A mismatch 488 (Not acceptable here) or 606 (Not acceptable)

A Warning header with warning code 304 (media type not available)

305 (incompatible media format)


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Negotiation of Media (2/3)INVITE sip:[email protected] SIP/2.0CSeq: 1 INVITEContent-Length: 183

Content-Type: application/sdpContent-Disposition: session

v=0o=collins 123456 001 IN IP4 station1.work.coms=c=IN IP4 station1.work.comt=0 0m=audio 4444 RTP/AVP 2 4 15a=rtpmap 2 G726-32/8000a=rtpmap 4 G723/8000a=rtpmap 15 G728/8000a=inactive

Daniel Boss

b

a

SIP/2.0 200 OK

CSeq: 1 INVITE

Content-Length: 157

Content-Type: application/sdp

Content-Disposition: session

v=0

o=coll ins 45678 001 IN IP4 station2.work.com

s=

c=IN IP4 station2.work .com

t=0 0

m=audio 6666 RTP/AVP 4 15

a=rtpmap 4 G723/8000

a=rtpmap 15 G728/8000a=inactive


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Daniel Boss

d

c

INVITE sip:[email protected] SIP/2.0CSeq: 2 INVITEContent-Length: 126Content-Type: application/sdpContent-Disposition: session

v=0o=collins 123456 002 IN IP4 station1.work.coms=c=IN IP4 station1.work.comt=0 0m=audio 4444 RTP/AVP 15

a=rtpmap 15 G728/8000

ACK sip:[email protected] SIP/2.0From: Daniel; tag = abcd1234To: Boss; tag = xyz789CSeq: 1 ACKContent-Length: 0

Negotiation of Media (3/3)


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Determine the capabilities of a potential called

party Accept Header

Indicate the type of information that the sender hopes toreceive

Allow Header Indicate the SIP methods that it can handle

Supported Header Indicate the SIP extensions that it can be supported

OPTIONS Method


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Daniel Boss

b

aOPTIONS sip:[email protected] SIP/2.0Via: SIP/2.0/UDP Station1.work.com; branch=z9hG4bK7890123From: Daniel; tag=lmnop123To: BossCall-ID: [email protected]: DanielCSeq: 1 OPTIONSAccept: application/sdpContent-Length: 0

SIP/2.0 200 OK

Via: SIP/2.0/UDP Station1.work.com; branch=z9hG4bK7890123

From: Daniel; tag=lmnop123

To: Boss; tag=xyz5678

Call-ID: [email protected]

CSeq: 1 OPTIONSAl low: INVITE, ACK, CANCEL, OPTIONS, BYE

Supported: newfield

Content-Length: 146

Content-Type: application/sdp

v=0

o=manager 45678 001 IN IP4 station2.work.com

s=c=IN IP4 station2.work.com

t=0 0

Usage of the OPTIONS Method


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SIP Extensions and Enhancements

SIP INFO Method

SIP Event Notification

SIP for Instant Messaging

SIP REFERMethod

Reliability of Provisional Responses

SIP UPDATE Method

Integration of SIP Signaling and Resource Management


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SIP Extensions and Enhancements

RFC 2543, March 1999

SIP has attracted enormous interest.

Traditional telecommunications companies, cable TV providersand ISP

A large number of extensions to SIP have been proposed.

SIP will be enhanced considerably before it becomes an Internetstandard.

RFC 3261

183 Session Progress

Supported header


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183 Session Progress

It has been included within the RFC 3261.

To open one-way audio path from called end to calling end From the called party to calling party

Enable in-band call progress information to be transmitted

Tones or announcements

Interworking with SS7 network ACM (Address Complete Message)

Ring back tone in 183 Session Progress


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The Supported Header The Require Header (client server)

A client indicates that a server must support certain extension withRequire header

420 bad extension with Unsupported header

The Supported header

Way 1:

Options (client server)

Response with Supported (server client)

Way 2:

Request with Supported (client

server) 421 extension required with Require (server client)


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SIP INFO Method

Specified in RFC 2976

For transferring information during an ongoing session Application-layer information could be transferred in the middle

of a call.

A powerful, flexible tool to support new services

The transfer of DTMF digits

The transfer of account balance information

Pre-paid service

The transfer of mid-call signaling information generated inanother network and passed to the IP network via a gateway


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SIP Event Notification

RFC 3265 hasaddressed the issue ofevent notification.

SUBSCRIBE andNOTIFY

Event header

Subscriber Notifier

SUBSCRIBE

200 OK

NOTIFY

200 OK

NOTIFY

200 OK

a

b

c

d

e

f

Current stateinformation

Updated stateinformation


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SIP for Instant Messaging (1/3)

The IETF working group SIP for Instant Messaging

and Presence Leveraging Extensions (SIMPLE) RFC 3428 Session Initiation Protocol (SIP)

Extension for Instant Messaging

A new SIP method MESSAGE This request carries the actual message in a message body.

A MESSAGE request does not establish a SIP dialog.


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DanielBoss sip:Server.work.com

MESSAGE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK7890Max-Forwards: 70From: BossTo: DanielCall-ID: [email protected]: 1 MESSAGEContent-Type: text/plainContent-Length: 19Content-Disposition: render

Hello. How are you?

MESSAGE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP server.work.com; branch=z9hG4bKxyz1Via: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK7890Max-Forwards: 69From: BossTo: DanielCall-ID: [email protected]: 1 MESSAGEContent-Type: text/plainContent-Length: 19Content-Disposition: render

Hello. How are you?

SIP/2.0 200 OKVia: SIP/2.0/UDP server.work.com; branch=z9hG4bKxyz1Via: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK7890From: BossTo: DanielCall-ID: [email protected]: 1 MESSAGE

Content-Length: 0

SIP/2.0 200 OKVia: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK7890From: BossTo: DanielCall-ID: [email protected]: 1 MESSAGEContent-Length: 0

a

b

c

d



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DanielBoss sip:Server.work.com

MESSAGE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP station1.work.com; branch=z9hG4bK123Max-Forwards: 70From: DanielTo: BossCall-ID: [email protected]: 1101 MESSAGE

Content-Type: text/plainContent-Length: 22Content-Disposition: render

Im fine. How are you?

MESSAGE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP server.work.com; branch=z9hG4bKabcdVia: SIP/2.0/UDP station1.work.com; branch=z9hG4bK123Max-Forwards: 69From: DanielTo: Boss

Call-ID: [email protected]: 1101 MESSAGEContent-Type: text/plainContent-Length: 22Content-Disposition: render

Im fine. How are you?

SIP/2.0 200 OKVia: SIP/2.0/UDP server.work.com; branch=z9hG4bKabcdVia: SIP/2.0/UDP station1.work.com; branch=z9hG4bK123From: DanielTo: BossCall-ID: [email protected]: 1101 MESSAGE

Content-Length: 0

SIP/2.0 200 OKVia: SIP/2.0/UDP station1.work.com; branch=z9hG4bK123From: DanielTo: BossCall-ID: [email protected]

CSeq: 1101 MESSAGEContent-Length: 0

e

f

g

h



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SIP REFER Method RFC 3515 The Session Initiation Protocol (SIP) Refer

Method

To enable the sender of the request to instruct the receiverto contact a third party With the contact details for the third party included within the REFER

request

For Call Transfer applications

The Refer-to: and Refer-by: Headers

The dialog between Mary and Joe remains established. Joe could return to the dialog after consultation with Susan.


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

REFER sip:[email protected] SIP/2.0Via: SIP/2.0/UDP station1.work.com; branch=z9hG4bK789Max-Forwards: 70From: Mary; tag=123456To: Joe; tag=67890Contact: MaryRefer-To: Susan

Call-ID: [email protected]: 123 REFERContent-Length: 0

SIP/2.0 202 AcceptedVia: SIP/2.0/UDP station1.work.com; branch=z9hG4bK789From: Mary; tag=123456To: Joe; tag=67890

Contact: JoeCall-ID: [email protected]: 123 REFERContent-Length: 0

INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP station2.work.com; branch=z9hG4bKxyz1Max-Forwards: 70

From: Joe; tag=abcxyzTo: SusanContact: JoeCall-ID: [email protected]: 567 INVITEContent-Type: application/sdpContent-Length: xx

Content-Disposition: session{message body}

a

b

c

Call Transfer (1/2)

sip:[email protected]

Call Transfer (2/2)


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sip:[email protected]:[email protected]

fg

SIP/2.0 200 OKVia: SIP/2.0/UDP station2.work.com; branch=z9hG4bKxyz1From: Joe; tag=abcxyzTo: Susan; tag=123xyzCall-ID: [email protected]: 567 INVITE

Content-Type: application/sdpContent-Length: xxContent-Disposition: session{message body}

ACK sip:[email protected] SIP/2.0Via: SIP/2.0/UDP station2.work.com; branch=z9hG4bKxyz1Max-Forwards: 70

From: Joe; tag=abcxyzTo: Susan; tag=123xyzCall-ID: [email protected]: 567 ACKContent-Length: 0

NOTIFY sip:[email protected] SIP/2.0Via: SIP/2.0/UDP station2.work.com; branch=z9hG4bK123Max-Forwards: 70

To: Joe; tag=67890From: Mary; tag=123456Contact: JoeCall-ID: [email protected]: 124 NOTIFYContent-Type: message/sipfrag;version=2.0Content-Length: 15

SIP/2.0 200 OK

SIP/2.0 200 OKVia: SIP/2.0/UDP station2.work.com; branch=z9hG4bK123To: Joe; tag=67890From: Mary; tag=123456Call-ID: [email protected]

CSeq: 124 NOTIFYContent-Length: 0

h

( )


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Reliability of Provisional Responses (1/2)

Provisional Responses

100 (trying), 180 (ringing), 183 (session in progress) Are not answered with anACK

If the messages is sent over UDP

Unreliable

Lost provisional response may cause problems wheninteroperating with other network

180 (ring), 183 (session progress) Q931 alerting or ISUPACM (Address Complete Message)

These external networks need these information to drive a statemachine.


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Reliability of Provisional Responses (2/2)

E.g., a call to an unassigned number

ACM to create a one-way path to relay an announcement suchas The number you have called has been changed

If the provisional response is lost, the caller might left in the darkand not understand why the call did not connect.

RFC 3262 Reliability of Provisional Responses in SIP

Response sequence (RSeq): a request header

Response ACK (RAck): a response header

Option tag=100rel with supported:/unsupported:

Provisional Response ACK (PRACK) method


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Provisional Response ACK (1/2)

INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP ClientA.network.com; branch=z9hG4bK7890123

Supported: 100relRequire: 100rel

From: sip:[email protected]; tag=lmnop123

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

CSeq: 1 INVITE

??SIP/2.0 180 Ringing


Require: 100relRSeq: 567890


To: sip:[email protected]; tag = xyz123Call-ID: [email protected]

CSeq: 1 INVITE

Response

Lost

b

a

c

ResponseRetransmit

[email protected] [email protected]

SIP/2.0 180 Ringing


Require: 100rel

Start timer T1

default=0.5 sec

Provisional Response ACK (2/2)[email protected] [email protected]


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SIP/2.0 180 Ringing

Via: SIP/2.0/UDP ClientA.network.com; branch=z9hG4bK7890123Require: 100rel

RSeq: 567891


To: sip:[email protected]; tag = xyz123


CSeq: 1 INVITE

d

c

eSIP/2.0 200 OK



To: sip:[email protected]; tag=xyz123


CSeq: 2 PRACK

PRACK sip:[email protected] SIP/2.0


RAck: 567891 1 INVITE


To: sip:[email protected]; tag=xyz123


CSeq: 2 PRACK

Th SIP UPDATE M th d


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The SIP UPDATE Method

RFC 3311 The Session Initiation Protocol (SIP)

UPDATE Method To enable the modification of session information

before a final response to an INVITE is received

E.g., to change the codec One important usage is when reserving network

resources as part of a SIP session establishment

Integration of SIP Signaling and

R M t (1/2)


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Resource Management (1/2) Ensuring that sufficient resources are available to

handle a media stream is a very important.

The signaling might take a different path from themedia.

Assume resource-reservation mechanisms are

available End-to-end status

End-to-end network resources are reserved as part of sessionestablishment.

Segmented status A certain amount of network resources are reserved in

advance for a certain type of usage.

Policing functions at the edge of the network


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Usage of SIP for Features and Services

Call Forwarding

Consultation Hold

U f SIP f F t /S i (1/2)


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Usage of SIP for Features/Services (1/2) Personal Mobility (Registrar)

Call-transfer application (with REFER method)

One number service through forking proxy

Call-completion services (by Retry-After: header)

Click-to-call applications or Web service (SIP address

is a URL) Existing supplementary services in traditional

telephony Call waiting, call forwarding, multi-party calling, call screening

Usage of SIP for Feat res/Ser ices (2/2)


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Usage of SIP for Features/Services (2/2) Proxy invokes various types of advanced

feature logic. Policy server (call-routing, QoS)

Feature server (subscriber-specific feature data, ex:screening lists, forwarding information)

Authentication server Use the services of an IN SCP over INAP

Use Open Service Access (OSA)

Call Forwarding


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Call Forwarding Call-forwarding-

on-busy Call-forwarding-

on-no-answer Timeout

CANCEL method

Call-forwardingunconditional


From: sip:user1

To: sip:[email protected]

Contact: User1

CSeq: 1 INVITE

SIP/2.0 100 Trying

From: sip:user1


CSeq: 1 INVITE


From: sip:user1


Contact: User1

CSeq: 1 INVITE

SIP/2.0 486 Busy Here

From: sip:user1


CSeq: 1 INVITE


From: sip:user1


CSeq: 2 INVITE

SIP/2.0 200 OK

From: sip:user1


Contact: sip:[email protected]

CSeq: 2 INVITE

SIP/2.0 200 OK

From: sip:user1


Contact: sip:[email protected]

CSeq: 1 INVITE

User1 sip:Server.work.com

User2

User3

Consultation Hold INVITE sip:User A@there SIP/2.0

User A User B User C

a


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Consultation Hold

User A asks User B a

question, and User Bneed to check with UserC for the correct answer.

User B could use the

REFER method totransfer the call to UserC.

A SIP UPDATE

INVITE sip:User A@there SIP/2.0From: sip:User B@hereCall-ID: 12345SDP Description

SIP/2.0 200 OKFrom: sip:User B@hereCall-ID: 12345SDP Description

ACK sip:User A@there SIP/2.0From: sip:User B@hereCall-ID: 12345

Conversation

UPDATE sip:User A@there SIP/2.0From: sip:User B@hereCall-ID: 12345

Session Descriptiona=inactive

SIP/2.0 200 OKFrom: sip:User B@hereCall-ID: 12345Session Descriptiona=inactive

On hold

UPDATE sip:User A@there SIP/2.0From: sip:User B@hereCall-ID: 12345Session Descriptiona=sendrecv

SIP/2.0 200 OKFrom: sip:User B@hereCall-ID: 12345

Session Descriptiona=sendrecv

User B calls User C, speaks, then hangs up

b

c

d

e

f

gh

i

j


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Interworking

PSTN Interworking

SIP to PSTN [email protected] Proxy.work.com PSTN switchNGW


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INVITE

183 (Session Progress)Session description

One-way audio

a

b

g

i

j

m

100 (Trying)INVITE

100 (Trying)

183 (Session Progress)Session description

IAM

ACM

One-way audio

ANM

200 (OK)Updated session description

200 (OK)Updated session description

ACK

ACK

Two-way audio Two-way audio

c

de

f

h

k

l

n

o

IAM (initial address message)

ACM (Address Complete Message)

ANM (Answer Message)

PSTN Interworking


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PSTN Interworking

RFC 3372 - Session Initiation Protocol for

Telephones (SIP-T): Context and Architectures PSTN SIP PSTN

RFC 3204 - MIME media types for ISUP and QSIGObjects

MIME media types

For ISUP

The whole issue of interworking with SS7 is

fundamental to the success of VoIP in the realworld.

Summary


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Summary

The future for signaling in VoIP networks

Simple, yet flexible Easier to implement

Fit well with the media gateway control protocols

SIP is the protocol of choice for the evolution ofthird-generation wireless networks. SIP-based mobile devices will become available

SIP-based network elements will be introduced withinmobile networks.


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End.

Documents

Sip（session intial protocol）