SIP, the IMS, and P2PSIP › upload › ims.pdf · Session Description Protocol (SDP) is the common body for session initiation. SIP runs on any transport protocol (UDP, TCP, TLS,

SIP, the IMS, and P2PSIP

[email protected]

2

Outline

� The protocol

– SIP

� Two ways (among others) to use SIP

– The IMS

– P2PSIP

SIP (Session Initiation Protocol)

4

� SIP is an application-level, end-to-end, client-server, extensible, text

based protocol.

� The design base was HTTP and SMTP

� SIP was originally used to establish, modify and terminate multimedia

sessions in the internet.

� SIP has evolved to be able to set-up a broad range of sessions:

– Multimedia (e.g., voice, video, etc)

– Gaming

– Presence and Instant Messaging

� SIP messages are either requests or responses.

� SIP messages carry zero or more “bodies”.

� Session Description Protocol (SDP) is the common body for session

initiation.

� SIP runs on any transport protocol (UDP, TCP, TLS, SCTP)

– The spec mandates UDP and TCP. Other transport protocols are

optional

SIP General Overview

Response

Request

Client Server

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History of SIP

� In February 1996, Handley and Schooler submitted an Internet Draft on Session Invitation Protocol, SIPv1, to the Internet Engineering Task Force

– Sipv1 used Session Description Protocol (SDP) to describe sessions– Text-based– UDP-based transport

� Also in February 1996, Henning Schulzrinne submitted an Internet draft specifying the Simple Conference Invitation Protocol (SCIP) to the IETF

– SCIP was based on HTTP and was thus text-based– Defined its own format for session descriptions– TCP-based transport

� Presentations of SIPv1 and SCIP at the 35th IETF meeting in March 1996

� In December 1996, a new draft on Session Initiation Protocol (SIPv2) was submitted by Handley, Schulzrinne and Schooler

– SIPv2 merged SIPv1 and SCIP– SIPv2 was based on HTTP, and was thus text-based– Used both UDP and TCP– Used SDP to describe multimedia sessions

� In December 1997, the decision was made to split SIP into a base spec and extensions

� In February 1999, SIP reached the proposed standard level and was published as RFC 2543

� In June 2002, the current SIP specification, RFC 3261, was published

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Overview of SIP Functionality

� SIP provides the following functionality:

– User location (not geographical location)

� Determination of the end system to be used for communication

– User availability

� Determination of the willingness of the other party to engage in

communications

– User capabilities

� Determination of the media parameters to be used

– Session set-up

� Establishment of session parameters at both called and calling

party

– Session management

� Transfer and termination of sessions, modifying session

parameters

� SIP does not provide services

– But it enables the system to provide services

– It has been demonstrated that it is easy to provide services with SIP

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SIP Logical Entities� User Agent (UA): An endpoint

– User Agent Client (UAC): sends requests, receives responses

– User Agent Server (UAS): receives requests, sends responses

� Proxy server: A network host that proxies requests and

responses, i.e., acts as a UAC and as a UAS.

� Redirect server: a UAS that redirects request to other

servers.

� Back-to-back User Agent: a UAS linked to a UAC– Acts as a UAS and as a UAC linked by some application logic

� Registrar: A special UAS that accepts only registrations

Response

Request

UAC Proxy

Response

Request

UAS

Response

Request

B2BUA

ResponseRequest

8

Stateless and Stateful SIP Proxies

� There are several types of SIP proxies, depending on the

state they keep:

� Stateless proxy: a proxy that does not keep any state when forwarding requests and responses.

– A simple message forwarder

� Transaction stateful proxy, or stateful proxy: a proxy that stores state during the duration of the transaction.

– Maintains a server transaction and a client transaction

� Call stateful proxy: a proxy that stores all the state pertaining to a session (e.g., from INVITE to BYE).

– A call stateful proxy is always a transaction stateful proxy, but not the other way round.

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The SIP Trapezoid

1. INVITE2. INVITE

Alice

atlanta.com biloxy.com

7. 180 (Ringing)

4. INVITE

6. 180 (Ringing)

3. 100 (Trying)5. 100 (Trying)

14. 200 (OK)

13. BYE

9. 200 (OK)

Bob

Bob is alerted

8. 180 (Ringing)Bob accepts

the session

10. 200 (OK)11. 200 (OK)

12. ACK

Media Session

Alice hangs up

Alice wishes to call Bob.

She sends an INVITE to

her proxy, atlanta.com

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

atlanta.com proxy

server locates the proxy

server at biloxy.com

biloxy.com proxy server

consults a location service

to find Bob’s IP address

100 Trying indicates that

proxy is working on behalf of

Alice to route the INVITE

Indicates that Bob’s

phone rings

Indicates that the call

has been answeredConfirms the reception of

the 200 OK final response.

ACK is sent directly to Bob,

bypassing the proxies

When Alice hangs up, a

BYE message is sent

To call Bob,

Alice uses his

SIP identity, a

type of URI

called a SIP

URI

Terminates the

session

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SIP Addresses

� SIP uses Uniform Resource Identifiers (URIs). At least, SIP

URIs and SIPS URIs are supported, although others (such as

TEL URL) are commonly supported.

– sip:[email protected]

– sips:[email protected]

– tel:+358-9-299-3283

– sip:proxy.atlanta.com:5060

– sip:another-proxy.biloxi.com;transport=UDP

� SIP and SIPS URIs must include a host name, and may

include username, may include port numbers, may include

parameters

� Non SIP/TEL URIs are also valid under certain circumstances:

HTTP, IM, PRES, MAILTO…

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An Example of a SIP Request

INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP [5555::aaa:bbb:ccc:ddd];branch=z9hG4bKnashds7Max-Forwards: 70Route: <sip:pcscf1.visited1.net;lr>, <sip:scscf1.home1.net;lr>From: <sip:[email protected]>;tag=171828To: <sip:[email protected]>Call-ID: cb03a0s09a2sdfglkj490333 Cseq: 127 INVITEContact: <sip:[5555::aaa:bbb:ccc:ddd]>Content-Type: application/sdp Content-Length: 248

v=0o=- 2987933615 2987933615 IN IP6 5555::aaa:bbb:ccc:ddds=-c=IN IP6 5555::aaa:bbb:ccc:ddd t=907165275 0m=audio 3458 RTP/AVP 97 96 0 15a=rtpmap:97 AMR a=fmtp:97 mode-set=0,2,5,7; maxframes=2a=rtpmap:96 G726-32/8000

Method

Request-URI

Version

Header

Body

Request

Line

Header Field

Header Field Name

Header Field Value

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SIP Methods� INVITE: create a session

� BYE: terminates a session

� ACK: acknowledges a final response for an INVITE request

� CANCEL: cancels an INVITE request

� REGISTER: binds a public SIP URI to a Contact address

� OPTIONS: queries a server for capabilities

� SUBSCRIBE: installs a subscription for a resource

� NOTIFY: informs about changes in the state of the resource

� MESSAGE: delivers an Instant Message

� REFER: used for call transfer, call diversion, etc.

� PRACK: acknowledges a provisional response for an INVITE request

� UPDATE: changes the media description (e.g. SDP) in an existing

session

� INFO: used to transport mid-session information

� PUBLISH: publication of presence information

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SIP Responses (1/2)

� SIP defines two types of responses– Final responses convey the result of the request processing, and are

sent reliably– Provisional responses provide information on the progress of the

request processing, but are not sent reliably in the core protocol (RFC 3261)

� Status codes ranges:– 100 – 199 Provisional (also known as informational responses)

� Server is performing some further action and does not yet have a definitive response

� Example: 180 Ringing – UA receiving the INVITE is trying to alert the user

– 200 – 299 Success� Request was successful� Example: 200 OK – the request has succeeded

– 300 – 399 Redirection� 3xx responses give information about the user’s new location or

about alternative services that might be available to satisfy the call

� Example: 302 Moved temporarily – retry the request at new address(es) specified in the Contact header

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SIP Responses (2/2)

� Status code ranges continued:– 400 – 499 Client error

� Definitive failure responses from particular server� Client should not retry the same request without

modification� Example: 401 Unauthorized – request requires user

authentication

– 500 – 599 Server error� Server itself is the cause of the error� Example: 500 Internal server error – server encountered

an unexpected condition

– 600 – 699 Global failure� Server has definitive information about a particular user� Example: 600 Busy everywhere – the callee is busy and

knows that no other end system will be able to accept the call

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SIP Routing and DNS System� SIP clients use DNS to route requests and find the next hop to route the

request

– By looking into a NAPTR (Naming Authority Pointer) record in DNS

– By looking into a SRV (Services) record in DNS

– By looking into A (IPv4) or AAAA (IPv6) records in DNS

� Example– Assumption: no transport and no port specified in the SIP URI

sip:[email protected]

NAPTR response: _sip._tcp.example.com

NAPTR query: example.com

SIP server DNS server

SRV query: _sip._tcp.example.com

SRV response: server1.example.com:5060

server2.example.com:5060

A or AAAA query: server1.example.com

A response: 10.8.2.1

SIP request sent to 10.8.2.1,

port 5060, using TCP

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SIP Registration

SIP server

and registrar

REGISTER sip:home1.net SIP/2.0

From: <sip:[email protected]>

To: <sip:[email protected]>

Contact: <sip:[email protected]>

Public user identity

sip:[email protected] is

bound to the contact address


SIP/2.0 200 OK



Contact: <sip: [email protected]>

1

2

play.home1.net

� The SIP registration function allows users to upload their current locations for use by proxy servers

– A REGISTER message associates a user’s SIP (or SIPS) URI with the

machine into which the user is currently logged

– The registrar writes this association into a database, from which it can be

fetched by a proxy server

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Routing: SIP Server in Proxy Mode

SIP server

and registrar

p1.home1.net

INVITE sip:[email protected] SIP/2.0




Record-Route: <sip:p1.home1.net;lr>

SIP/2.0 200 OK


Record-Route: <sip:p1.home1.net;lr>

2

3





1

SIP/2.0 200 OK

Contact: <sip:[email protected]>Record-Route: <sip:p1.home1.net;lr>

4

ACK sip:[email protected] SIP/2.0

Route: <sip:p1.home1.net;lr>5


play.home1.netpc33.example.net

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Routing: SIP Server in Redirect

Mode

SIP server

and registrarINVITE sip:[email protected] SIP/2.0




SIP/2.0 200 OK


4

5





1

pc33.example.net

SIP/2.0 302 Moved Temporarily

Contact: <sip:[email protected]>2



play.home1.net

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Forking Proxies

� A proxy server can send an INVITE to a number of locations at the same time

– This type of parallel search is known as forking

� A proxy can route messages in parallel or in sequence– In parallel forking, all locations are attempted simultaneously

– In sequential forking, the proxy tries different locations one after the other

� Forking may result in multiple 2xx responses– Each will establish a dialog

Forking proxy

Sip:[email protected]



(1) INVITE

(2) INVI

TE

(3) INVITE

The IMS

(IP Multimedia Subsystem)

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What is IMS?

• IP Multimedia Subsystem (IMS) is an

architecture for offering services on the

packet domain.

• Not just Voice over IP - IMS is a generic

architecture for offering multimedia

services.

• IMS is access agnostic, hence it is not

just for UMTS or GPRS, but will also

support WLAN, fixed line, etc.

IMSIMS

Multi Access

Packet Services

Rapid Services

IP Optimised

Mobile WorldInternet World

Technology

convergence

Increase

ARPU

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IMS architecture

RNC

P-CSCF

I-CSCF

MRF

MGW

MGCF

IMSIMS

S-CSCF

MSC(Server)SGSN

GGSN

CNCN

MGW

BSC

UMTS/GPRSUMTS/GPRS

WLAN / WLAN / WiMAXWiMAX

Fixed AccessFixed Access

Corporate / HomeCorporate / Home

SIP Application

ServersSIP Application

Servers

HSS

CDMA 2000CDMA 2000

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Terminal Terminal

Replicated - not reusable

Terminal Terminal

Common Functions

Routing & Discovery

Horizontal benefits of IMS

Vertically integrated services Horizontal integration

Common Functions

Routing & discovery

Application Logic

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Terminal Terminal


Terminal Terminal

Common Functions

Routing & Discovery

Horizontal benefits of IMSVertically integrated services Routing & discovery

Common Functions

Routing & discovery

Application Logic

Numbering, Identification, Addressing

NNI

Routing

Service invocation

Authentication

SIP Reach ability & Capability negotiation

Registration

Re-Usable Components

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Terminal Terminal


Terminal Terminal

Common Functions

Routing & Discovery

Horizontal benefits of IMSVertically integrated services Common Functions

Common Functions

Routing & discovery

Application Logic

Numbering, Identification, Addressing

NNI

Routing

Service invocation

Authentication

SIP Reach ability & Capability negotiation

Registration

Re-Usable Components

Common Directory

Common Deployment

Common Charging

Common Conferencing

Common Provisioning and O&M

Common Group / List Management

Common Presence

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Advertising

Cross-selling

On demandVotingSharing

IMS Control for all Broadband Services

Control &

Connectivity

Service

Access

IMS Service Control

Ericsson IMS supports convergence

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The IMS Eco System

• Operators

• Internet Providers

• Service Developers

• Content Providers

• Terminal Vendors

• Network Vendors

Telecom

Internet

Eco System

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IMS Poker

Poker Application

Presence MessagingPush to Talk

Poker Application

PresenceMessaging Push to Talk

Other IMS operators

NNI

Messages (Game)

API API

UNIUNI

Presence

PoC Voice

Messaging

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Architecture Elements

IM-SSF

I-CSCF

MRF

MGW

MGCF

HSS

CSE(SCP) SIP Application

Servers

OSA-SCS

SGW

OSA Application

Server

S-CSCFP-CSCF

BGFC

Database Elements

• HSS (Home Subscriber Server)

• SLF (Subscription Locator

Function)

IMS Control Elements

• S-CSCF

• P-CSCF

• I-CSCF

Control Plane Interworking

Elements

• MGCF - Media Gateway Control

Function

• BGCF - Breakout Gateway Control

Function

• SGW - Signaling Gateway

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Other Elements

IMS Service Elements

• AS (Application Server)

External Service and Service

Interworking Elements

• OSA SCS

• OSA Framework

• OSA Application Server

• CAMEL elements - IM-SSF (IP Multimedia

Switching Service Function)

Resource Elements

• Media Resources Function (MRF)

Media Interworking Elements

• MGW (Media Gateway)

IM-SSF

I-CSCF

MRF

MGW

MGCF

HSS

CSE(SCP) SIP Application

Servers

OSA-SCS

SGW

OSA Application

Server

S-CSCFP-CSCF

BGFC

Steps:

1) Route

signalling to

A’s home

network

2) Route

signalling to

B’s home

network

3) Route

signalling to B

terminal

P-CSCFP-CSCF

S-CSCFS-CSCF

Home A

P-CSCFP-CSCFVisited A

Home B

I-CSCFI-CSCF

HSSHSS

I-CSCFI-CSCF

14

15

17

19

20 1

2

3

Visited B

78

9 6

12

13

S-CSCFS-CSCF

SGSNSGSN

Radio Access NetworkRadio Access Network

GGSNGGSN GGSNGGSN

SGSNSGSN

Radio Access NetworkRadio Access Network

ASASASAS

451011

16

User plane (RTP)

18

SESSION ESTABLISHMENT

MODEL

32

PoC Architecture

SIP/IP Core

(IMS Based)

Media Plane

Home PoC Server

Control

Plane

ISC

RTP/RTCP

� The PoC server acts as an application server– Implements the ISC (SIP-based) interface

– It may be distributed (control and media planes)

P2PSIP

(Peer-to-peer SIP)

34

Background

� P2P systems are already being used to share files, distribute content, and establish multimedia sessions

� They mainly provide– Signalling level: rendezvous

– Media level: NAT traversal

� P2P systems are supposed to reduce the cost involved in managing and operating a network

� The P2PSIP working group has been chartered at the IETF

– To use a P2P system to establish multimedia sessions� To do in a standard way what Skype does in a

proprietary way

– It will produce a standard solution based on SIP and DHTs

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Traditional SIP Network

SIP

UA

SIP

UA

SIP Proxy SIP Proxy

Alice Bob

SIP Registrar SIP Registrar

first.com second.comSIP

UA

SIP

UA

SIP

UA

SIP

UA

36

SIP RegistrarSIP Registrar

SIP ProxySIP Proxy SIP ProxySIP Proxy

SIP Registrar SIP RegistrarSIP RegistrarSIP Registrar

SIP Proxy SIP Proxy

P2PSIP Network

first.com second.com

SIP

UA

SIP

UA

Alice Bob

SIP

UA

SIP

UA

SIP

UA

SIP

UA

37

Terminology

� Peer– They collectively run a distributed database algorithm

� Peer protocol (which is an overlay maintenance

protocol)

� Client– A client is a less capable peer

� It may not store records

� It may not participate in the maintenance of the

overlay

� It may not do either

– A client has one of more associated peers

– Communicates with peers using a client protocol

38

Peers, Clients, and SIP Entities

� Most peers and clients will be colocated with a SIP

entity– SIP UA, proxy, aplication server...

� But some may not be– Supernodes with no SIP functionality

39

Work at the IETF (1/3)In Scope

� Define a peer protocol:– Selection of P2P algorithm (likely a DHT)

� Define a client protocol.

� Protocols need to support NATs.

� Produce an architecture document:– Define how to create systems

– Media relays

� Also instant messaging and presence are in scope.

Source: P2PSIP

WG charter

40

Work at the IETF (2/3)Out of Scope

� The modification of the baseline SIP behavior �

Extensions handled in SIP WG.

� Not to develop new NAT traversal mechanism, if existing

ones can be used.

� Any research-like questions, e.g. how to assure unique

user id.

� To use something else than URIs for addressing

users/resources.

Source: P2PSIP

WG charter

41

P2PSIP Overlay

P2PSIP Bootstrap

Peer

P2PSIP Peer

P2PSIP Peer

P2PSIP Client

NAT

NAT

NAT

P2PSIP Overlay

<Resource-ID>

P2PSIP Resource

Record

Interacts with the P2PSIP

overlay through its associated

peer using the Client Protocol.

Does not run the distributed

database algorithm.

Participates in the P2PSIP

overlay and provides storage

and transport services to other

nodes.

Client Protocol

<Peer-ID>

First point of contact for a peer

joining the overlay. Can be located:

- By remembering peers from the

last time the peer was in the overlay

- Through multicast discovery

- Through manual configuration

- By contacting a bootstrap server

Peer ProtocolThe protocol spoken between

P2PSIP Peers to share

information and organize the

P2PSIP Overlay network.

Current understanding is that

SIP will not be used as the

Peer Protocol (SIP will be used

to initiated calls in the overlay

which has been organized

using the peer protocol).

The protocol spoken

between clients and

peers

Joining peer is a node

attempting to become a

P2PSIP Peer.

Admitting peer helps the joining

peer join the network.

42

P2PSIP Operations (1/2)

� P2PSIP peers are capable of performing different operations suchas:

– Joining and leaving– Get and put– Storing information on behalf of the overlay– Transporting messages

� To join a P2PSIP overlay, a joining peer needs to:– Obtain a Peer-ID and optionally a set of credentials

� The Peer-ID is a hash of a unique node identifier such as an IP address

– Locate a bootstrap peer for the overlay� The bootstrap peer will refer the joining peer to an

admitting peer– Contact an admitting peer

� The admitting peer will help the joining peer learn about other peers in the overlay and establish connections to them as appropriate

43

P2PSIP Operations (2/2)

� To perform a user registration (i.e. to insert the user’s contact

information into the overlay), a user needs to:

– Calculate a hash of her user name (e.g. [email protected]) to

produce a Resource-ID

– Locate the peer that is responsible for that Resource-ID

– Store a Resource-ID to contact address mapping in the

responsible peer

� To initiate a call:

– Calculate a hash of the callee’s user name to produce a

Resource-ID

� hash([email protected]) = 32B4A7F02C

– Locate the peer that is responsible for that Resource-ID in the

P2PSIP overlay

� A P2PSIP Resource Record with contact information is

obtained: [email protected] → 192.0.2.1:1234

– Send a SIP INVITE request to the callee

44

Vision for P2PSIP

� Some enterprises will use P2PSIP for their

communication systems, mainly to reduce their OPEX

� An Internet-wide P2PSIP network will exist. A standard

version of what Skype is today

� P2PSIP will be used between infrastructure nodes in

some architectures

� The peer protocol used by P2PSIP will be used to build

standards-based overlays that will not use SIP

45

Greatest Obstacles to P2PSIP

� Security and identity assertion– Is there a need for centralized components after all?

� Will the P2PSIP peer protocol be general enough to

support a large set of present and future DHTs?– Will we face interoperability problems?

� Performance– How much worse than a well-designed server-based

architecture?

� Battery and bandwidth constraint terminals– Will the use of a client protocol resolve this issue?

� Regulatory issues– Emergency calls, legal intercept, etc.

46

Documents

SIP, the IMS, and P2PSIP › upload › ims.pdf · Session Description Protocol (SDP) is the common body for session initiation. SIP runs on any transport protocol (UDP, TCP, TLS,