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IP Multimedia Subsystem Part 8 – IMS/SIP Project Marek Średniawa [email protected] Institute of Telecommunications Project is co-financed by European Union within the European Social Fund EIMS - 2015 1

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Page 1: IP Multimedia Subsystem Part 8 IMS/SIP Projectmareks/eims/EIMS-8-color.pdf · 2015-01-21 · IP Multimedia Subsystem Part 8 –IMS/SIP Project ... – of NGN/IMS based services/applications

IP Multimedia SubsystemPart 8 – IMS/SIP Project

Marek Ś[email protected]

Institute of Telecommunications

Project is co-financed by European Union within the European Social FundEIMS - 2015

1

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Types of projects

• Analytical – prepare a survey– of NGN/IMS based services/applications offered by a selected

operator

– of selected NGN/IMS service platforms and their capabilities

– of APIs and SDKs by operators for application development

• Programming– develop and configure a simple IMS/SIP service/application using

one of the recommended available tools and shareware or temporary license based software packages/solutions

EIMS - 2015

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Sample analytical projects

• Convergent services in a selected country (e.g.: UK, Germany, Czech Republic, Russia, India, France, Spain, Netherlands, Belgium, Italy, …).

• Convergent services offered by a selected operator, e.g.: – BT, Orange, Telecom Italia, O2, Vodafone, Telefonica, DT, KPN, FT

• Analysis of Service Delivery Platforms– Alcatel-Lucent, Nokia-Siemens Networks, Aepona, Cisco, Ericsson, HP,

ZTE, ….

• Architecture of BT21CN

• Architecture of Telecom Italia NGN2 network

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Sample programming projects

• FPH/SPL/UAN – call centre with ACD (automatic call distribution)

• Family and friends freephone

• Televoting – different variants

• CLV (Calling Line Verification)

• Personal Number

• Incoming call management

• Location based services

• Context sensitive click to call service

• Redirection of call to a website

• Own proposal – subject to final acceptance by the Lecturer

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Application Servers and trigerring

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IMS AS Functional Model

Home Subscriber Server

Serving-CSCF

Application Logic

Incoming Leg Control Model

(AS-ILCM)

Outgoing Leg Control Model

(AS-OLCM)

<<Sh>>(Diameter)

<<ISC>>(SIP)

<<ISC>>(SIP)

SIP Application Server

3GPP “call leg” = SIP “dialog” !!!

EIMS - 2015

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Until the Application

Server receives the

BYE-request, it will

send a MESSAGE-

request with the

“Hello World”

message every 30th

second

Sample IMS service – “Hello World”

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Software tools for implementation of IMS AS

• IMS – software platform and programming language neutral

• Sample open source approaches to implementation of a SIP

AS based on the Java Platform:

– JSR 116 – SIP Servlet API

• Sailfin - open source implementation on top of Glassfish

– JSR022 – JAIN™ Service Logic and Execution Environment (SLEE) API

Specification

• Mobicents

• In addition to the open source alternatives, there are many

commercial alternatives available.

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SIP Servlet API (SSA)

• The SIP Servlet specification based on HTTP Servlet API JSR053

– e.g., implemented by Apache Tomcat, Glassfish, and BEA WebLogic

• SIP Servlet

– A Java-based application component managed by a SIP servlet container performing SIP signaling

– platform independent Java classes compiled to platform neutral code that can be loaded dynamically and run by a java-enabled SIP application server

– Containers are server extensions that provide servlet functionality

– Servlets interact with SIP clients by exchanging request and response messages via the servlet container

• SIP Servlet Container

– part of an application server which provides network services over which requests and responses are received and sent

– manages the network listen points on which it listens for incoming SIP traffic (listen point: a combination of transport protocol, IP address and port number)

– decides which applications to invoke and in what order

– A servlet container contains and manages servlets during their lifecycle

• SIP Servlet API v1.0 - JSR116

• SIP Servlet API v1.1 - JSR289

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JSR116 - SIP Servlet API: SIP Servlet Lifecycle

Creatednew()

Initialized

Destroyed

init() destroy()

service()

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Application Triggering

• Selecting by the S-CSCF of an appropriate Application Server– The “Application Server Subscription Information” is the set of all

Filter Criteria that are stored within HSS for service profile for a specific user

• Sent by the HSS to the S-CSCF via the Cx Interface during registration - REGISTER request

• More then one set of Filter Criteria may be sent if the user identity belongs to different service profiles

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Filtering Criteria

• Filtering done for initial SIP request messages– “initial Filter Criteria” - iFC

• The S-CSCF applies filter criteria to determine the need to forward SIP requests to ASs

• Filter criteria defined in XML– According to 3GPP TS 29.228

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Content of iFC

AS address: Address to be used to access the AS for a particular subscriber.

Default handling: Indicates whether to abandon matching of lower priority triggers and to release the dialogue, or to continue the dialogue and trigger matching.

Trigger point: Defines the Service Point Trigger (SPT) for the application. If there is a match between the SPT and the request, then the S-CSCF proxies the request to the Application Server address.

iFC Priority: If there are multiple iFCs assigned to a subscriber, the priority determines the order in which the S-CSCF processes them.

Service Information: Optional. If present, the S-CSCF includes it in the body of the SIP request which is sent to the AS (only included by the S-CSCF in REGISTER-requests if the S-CSCF acts as a User Agent Client (UAC).

Trigger Point

ConditionTypeCNF : boolean

Application Server

ServerName : SIP URLDefault Handling: enumerated

Initial Filter Criteria

Priority: integer

Service Information

ServiceInfo : string

0...11...n

Service Point Trigger

ConditionNeg ated: booleanGroup: list of integer

0..1

ProfilePartIndicator : enumerated

Trigger Point

ConditionTypeCNF : boolean

Application Server

ServerName : SIP URLDefault Handling: enumerated

Initial Filter Criteria

Priority: integer

Service Information

ServiceInfo : string

0...11...n

Service Point Trigger

ConditionNeg ated: booleanGroup: list of integer

0..1

ProfilePartIndicator : enumerated

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Trigger Point - TP

• TP describes the trigger points that should be analyzedto determine whether the indicated AS should be contacted or not

• A TP expression is built of atomic expressions (i.e. Service Point Trigger) linked by Boolean operators AND, OR and NOT

• Absence of a TP instance indicates unconditional triggering to an AS

• Individual SPT statements can also be negated

• A Service Point Trigger can use the following information:

– Request-URI

– SIP Method

– SIP Header

– Session case (Values: {originating, terminating – registered, terminating –unregistered, originating –unregistered}

– Session description information (line and content of the SDP body)

ConditionNegated : Boolean

Group: list of integer

RegistrationType: list of enumerated

Service Point Trigger

RequestURI : string

Request-URI

Method : string

SIP Method

Header : string

Content: string

SIP Header

SessionCase : enumerated

Session Case

Line : string

Content : string

Session

Description

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Service Profile ExampleSIP Requests with the method “MESSAGE” or if the method is

“SUBSCRIBE” and the From-header is not “joe”

– Method = "MESSAGE" OR (Method="SUBSCRIBE" AND NOT Header = "from" Content = "joe")

<?xml version="1.0" encoding="UTF-8"?>

<IMSSubscription

xmlns:xsi="http://www.w3.org/2001/XMLSchema-

instance" xsi:noNamespaceSchemaLocation="D:\

\CxDataType.xsd">

<PrivateID>[email protected]</PrivateID>

<ServiceProfile>

<PublicIdentity>

<BarringIndication>1</BarringIndication>

<Identity> sip:[email protected] </Identity>

</PublicIdentity>

<PublicIdentity>

<Identity> sip:[email protected] </Identity>

</PublicIdentity>

<InitialFilterCriteria>

<Priority>0</Priority>

<TriggerPoint>

<ConditionTypeCNF>1</ConditionTypeCNF>

<SPT>

<ConditionNegated>0</ConditionNegated>

<Group>0</Group>

<Method>MESSAGE</Method>

</SPT>

<SPT>

<ConditionNegated>0</ConditionNegated>

<Group>0</Group>

<Method>SUBSCRIBE</Method>

</SPT>

<SPT>

<ConditionNegated>1</ConditionNegated>

<Group>1</Group>

<SIPHeader>

<Header>From</Header>

<Content>"joe"</Content>

</SIPHeader>

</SPT>

<SPT>

<ConditionNegated>0</ConditionNegated>

<Group>1</Group>

<Method>MESSAGE</Method>

</SPT>

</TriggerPoint>

<ApplicationServer>

<ServerName>sip:[email protected]</ServerName

>

<DefaultHandling>0</DefaultHandling>

</ApplicationServer>

</InitialFilterCriteria>

</ServiceProfile>

</IMSSubscription>

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Operation modes of AS

• Modes of AS operation for processing SIP requests:– AS acting as terminating UA, or redirect server

– AS acting as originating UA

– AS acting as a SIP proxy

– AS performing third party call control - 3PCC/B2BUA mode

– AS not involved or no longer involved

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AS acting as terminating UA/redirect server

• The incoming SIP request is proxied by the S-CSCF to the AS

• The AS acts as:

– a normal User Agent (UA), or

– a Redirect Server • according to RFC 3261, Section 8.3

Serving-CSCF

Application Server

SIP dialog

#1

SIP dialog

#1

From: X

To: Y

Call-ID: Z

From: X

To: Y

Call-ID: Z

EIMS - 2015

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AS acting as originating UA

• AS generates a SIP Request and sends to the S-CSCF, which then proxies it towards the destination

• Usually used in case of triggers or state changes in the service logic.

– Timers

– Change of the presence status

– Change of location

– etc.

Serving-CSCF

Application Server

SIP dialog

#1

SIP dialog

#1

From: X

To: Y

Call-ID: Z

From: X

To: Y

Call-ID: Z

EIMS - 2015

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AS acting as a SIP proxy

• Incoming SIP Request is proxiedby the S-CSCF to the AS which then proxies it back to the S-CSCF

• The S-CSCF then proxies it towards the destination

• During the proxy operation the AS can:

– add,

– remove, or

– modify

the header contents of the SIP request, according to the Proxy server operation rules RFC3261

SIP dialog

#1

From: X

To: Y

Call-ID: Z

Serving-CSCF

Application Server

SIP dialog

#1

SIP dialog

#1

From: X

To: Y

Call-ID: Z

From: X

To: Y

Call-ID: Z

SIP dialog

#1

From: X

To: Y

Call-ID: Z

EIMS - 2015

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AS – 3PCC/B2BUA mode• “3PCC refers to the ability of one entity to create a call in which communication is actually between other

parties.” [RFC3725]

• AS performing 3PCC acts as a Back-to-Back UA - B2BUA.

– B2BUA means that the AS acts as both a User Agent Client (UAC) and a User Agent Server (UAS) at the same time.

• Many types of 3PCC, e.g.:

– Routing B2BUA

– Initiating B2BUA

• The following slides show these two scenarios

SIP dialog

#1

From: X

To: Y

Call-ID: Z

Service Logic

SIP User Agent Server

(UAS)

SIP User Agent Client

(UAC)

SIP dialog

#2

From: P

To: Q

Call-ID: R

EIMS - 2015

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3pcc – Routing B2BUA mode

• The incoming SIP Request is proxied by the S-CSCF to the AS

• The AS then generates a new SIP request for a different SIP dialog, which it sends to the S-CSCF which the proxies it towards the destination.

SIP dialog

#2

From: P

To: Q

Call-ID: R

Serving-CSCF

Application Server

SIP dialog

#1

SIP dialog

#1

From: X

To: Y

Call-ID: Z

From: X

To: Y

Call-ID: Z

SIP dialog

#2

From: P

To: Q

Call-ID: R

EIMS - 2015

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3PCC – Initiating B2BUA mode

• AS initiates two requests, with different SIP dialogs.

• AS is responsible for correlating the two dialogs.

SIP dialog

#2

From: P

To: Q

Call-ID: R

Serving-CSCF

Application Server

SIP dialog

#1

SIP dialog

#1

From: X

To: Y

Call-ID: Z

From: X

To: Y

Call-ID: Z

SIP dialog

#2

From: P

To: Q

Call-ID: R

EIMS - 2015

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AS not involved or no longer involved

• Either the AS was never involved in the session, or during the session it has determined that it does not need to be involved anymore

• The AS controls its involvement using the Record-Route header

• If the AS does not insert itself in a Record-Route header, then this mode is used for all subsequent requests related to this SIP dialog.

Serving-CSCF

SIP dialog

#1

From: X

To: Y

Call-ID: Z

SIP dialog

#1

From: X

To: Y

Call-ID: Z

EIMS - 2015

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A conceptual model for IMS service development

• The service uses the service enablers to provide some functionality.

• Sample service enablers:

– Charging

– Presence

– Voice calling

– Location

– Messaging

– Device management

– Over The Air (OTA) delivery

• The service is accessed through a service access point:

– SIP - ISC reference point in IMS

– HTTP Ut reference point in IMS

Service

Service Enablers

Service access points

EIMS - 2015

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Using standardized IMS enablers and client API’s

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Device Java IMS API’s (JCP defined)

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JSR-281 architecture

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Recommended software tools

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Ericsson Service Design Studio

http://devtools.ericsson.com/

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Service Development Studio (SDS)

• Ericsson's developer tool runs on a PC and is freely available and downloadable for developers to enable them to rapidly design and test their IMS applications– Download available at: ftp://ftp.ericsson.ca/

• SDS provides:– high-level Client-side IMS Core

– Communication Services APIs conforming to JSR 281,

– server APIs conforming to JSR 116 and JSR 289

– device-, IMS core-, and IMS service enabler simulators

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SDS Overview

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SDS capabilities

• Client-Server IMS application development and debugging– Complete IDE based on Eclipse, with support for Web Tools Platform (WTP) and Wireless ToolKit (WTK)

– Sun, Sony Ericsson and Nokia WTK

• End end-to-end testing with emulated IMS network– IMS core: CSCF, HSS, DNS

– Presence and groups, PGM emulator

– Push-to-Talk, PTT-AS emulator

– IMS Messaging IMS-M emulator

– Real-time content and IPTV core emulator (basic)

– Voice over IP P-2-P (VoIP as part of 3GPP MMTel in roadmap)

• Support for Mobile and Fixed access and Devices:– Standard JavaME Mobile, Fixed Broadband and WLAN accesses

– phones for an IMS service subset

• Support for JavaEE/SIP Target Servers– Open Source Glassfish/SailFin as default in SDS

– Sun SailFin AS verified as target server

– Sun SGCS 1.5 verified as target server

– Oracle OCCAS 4.0verified as target server

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SDS components

• Design Environment, Eclipse IDE (WTP, Eclipse ME, WTK)

• Service API’s

• Automatic Test Framework, Test agent

• Visual Traffic Flow (VTF) tool

• Visual Network

• IMS Java Client Utility (IJCU)

• Device Emulator (Sun, SEMC, Nokia WTK)

• Server AS (Glassfish/SailFin)

• IMS Core Emulator (CSCF, HSS, DNS)

• Presence and Group mgt Emulator (PGM)

• Push-to-Talk Emulator (PTT)

• IMS Messaging Emulator (IMS-M)

• IPTV server emulator (basic)

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Ericsson Developers Tools

• JSR 281

– Enables application programmers to easily develop applications that can integrate with the IP Multimedia Subsystem (IMS)

– http://devtools.ericsson.com/jsr281/overview

• JSR 325

– Provides Java ME based devices effortless access to a set of essential IMS Communication Enablers

– http://devtools.ericsson.com/jsr325/overview

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Ericsson Developers Tools – JSR 281

• Standard API that abstracts the underlying IMS technology and allows development ofadvanced convergent multimedia services using Java

• Defines a set of high-level functions enabling Java ME applications to access IMS functionality:

– High-level support for the IMS registration mechanism

– Support for co-location of multiple IMS Services

– Use of IMS service sessions (based on SIP sessions)

– Use of media connections

– Addressing Quality of Service

– Hiding and encapsulating internal protocols managed and used by the IMS protocol stack

• The authentication mechanism used by IMS is hidden from application

• The specification is available for download at the JCP (Java Community Process) page for JSR 281

– http://www.jcp.org/en/jsr/detail?id=281,

• Reference implementation – send request to: [email protected]

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Ericsson Developers Tools – JSR 325

• JSR325 defines a high abstraction level API to facilitate theimplementation of some standardized communicationservices on the terminal side.

• JSR325 defines APIs for three different IMS CommunicationEnablers:– Presence

– Instant Messaging (IM)

– XML Document Management (XDM)

• The Reference Implementation is available as open source under Apache version 2.0. RI source code can be found at– http://devtools.ericsson.com/node/23

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Aepona Gbox / Xway

http://downloads.aepona.com/gboxweb/gboxdownload.php

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Aepona Gbox/Xway

• A plug-in for Eclipse aimed to assist the application developers of Appium-Xway

• GBox simplifies the application development process for applications running on Appium-XWay by providing tools and wizards

• Enables an average Java developer with basic telecom knowledge to create easily advanced communication applications with the help of off-the-shelf Java development tools

• Runs on a standard desktop or laptop PC

• Provides an embedded service execution environment and simulation and testing tools with graphical user interfaces that allow verification and testing of applications.

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Aepona Gbox/Xway

• Appium-Xway

– service execution environment for managing and delivering telecom services

• Download available at: • http://downloads.aepona.com/gboxweb/gboxdownload.php

– Prior registration and license key from Aepona required

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Gbox/Xway SW/HW requirements

• SW requirements

– Prior to installing Appium-Gbox installation of thefollowing software components is required:

• Java, Standard Edition 5.0 (JSE) from http://java.sun.com.

• Eclipse 3.2 from http://www.eclipse.org

• Appium-XWay 4.0.2 or later

• HW requirements:

– PC/laptop with 1 GHz, 1 GB RAM, 1 GB free HDD, Windows XP or Linux

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Functional Model of XWay

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Xway modules

• Web services (including Parlay X)– Enables development and deployment of Web services and includes

Web service exposure functions such as SLA, Provisioning, Load Control, Load Balancing and Security.

• SIP / IMS– Enables development and deployment of JAIN SIP APIs.

• OSA/Parlay– Enables development and deployment of OSA/Parlay services.

• OMA– Enables development and deployment of OMA applications.

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Gbox testing tools

• A network simulator

– can be used to test application in the development environment

• Logging and Tracing capabilities

– enables you to view the dynamic behaviourof the system

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Open Source Core IMS

Fraunhofer Institute FOKUS, Germany

www.openimscore.org

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Open Source Core IMS

• Open Source implementation of:

– IMS Call Session Control Functions (CSCFs)

– lightweight Home Subscriber Server (HSS)

which together form the core elements of all IMS /NGN architectures as specified by 3GPP, 3GPP2, ETSI TISPAN and the PacketCable intiative

• The components are all based upon Open Source software:

– e.g. the SIP Express Router (SER) or MySQL

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Source: http://www.openimscore.org/

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Open Source Core IMS architecture

• Home Subscriber Server– GNU General Public License v.2

• 3GPP compliant IMS Call Session Control Functions CSCFs)– Proxy-CSCF

– Interrogating-CSCF

– Serving-CSCF

– GNU General Public License v.2

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Source: http://www.openimscore.org/

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IMS Testbed based on open source components

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Source:The UCT IMS Client, http://uctimsclient.berlios.de/

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Open source 3GPP IMS/RCS framework

Doubango

http://www.doubango.org/index.html

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Doubango:open source 3GPP IMS/RCS framework

• Client-side components– Boghe - IMS/RCS Client for Windows

– IMSDroid - SIP/IMS Client for Android

– Idoubs -SIP/IMS VideoPhone for iOS (iPhone, iPad and iPod Touch)

• Server-side components– OpenVCS (Open Source Video Conferencing) - Server used to manage

Multipoint Control Units (MCU)

– Flash2IMS - Adobe® Flash® to SIP/IMS Gateway

• Download at: – http://www.doubango.org/index.html

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Doubango - conformance to standards

• SIP: RFC 3261, 3GPP TS 24.229 Rel-9

• IMS-AKA: RFC 3310, 3GPP TS 33.203, 35.205, 35.206, 35.207, 35.208, 35.909)

• Private extension headers for 3GPP

• IPv4/IPv6 dual stack

• UDP, TCP, TLS and SCTP

• Service-Route Discovery (RFC 3608)

• Proxy-CSCF discovery usingDHCPv4/v6 or/and DNS NAPTR

• SigComp (RFC 3320, 3485, 4077, 4464, 4465, 4896, 5049, 5112 , 1951)

• IPSec

• Security Agreement (RFC 3329)

• NAT Traversal (STUN, TURN and ICE)

• Preconditions (RFC 3312, 4032 and 5027)

• SMS over IP (3GPP TS 23.038, 24.040, 24.011, 24.341 and 24.451)

• ENUM (RFC 3761)

• The tel URI for Telephone Numbers (RFC 3966)

• SIP SIMPLE (Presence Subscription / Publication and Pager Mode IM)

• MMTel (UNI)

• SDP (RFC 3264)

• RTP/RTCP

• XCAP (RFC 4825)

• MSRP (RFC 4975 and 5547)

• Audio/Video (AMR, Speex, G.711, GSM, H.264, H.263, Theora...)

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The IMS Project for AndroidT-Mobile USA & Movial

51https://code.google.com/p/the-ims-open-source-project-for-android/EIMS - 2015

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Project Clearwater - IMS in the Cloud

52http://www.projectclearwater.org/

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Free and open source SIP servers

• Asterisk

• Cipango SipServlets 1.1 application server

• FreeSWITCH

• Kamailio, maintaining OpenSER

• Mobicents Platform (JSLEE[2] 1.0 compliant and Sip Servlets 1.1 compliantapplication server)

• Mysipswitch

• OpenSER

• OpenSIPS, fork of OpenSER

• SailFin

• SIP Express Router (SER)

• sipX

• YXA – a SIP server based on Erlang

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Free and/or open source software SIP clients

• Blink, a SIP client featuring IM, File Transfer and Desktop Sharing for MacOSX, Windows and Linux, GPL

• Ekiga, formerly known as GnomeMeeting, GPL

• Empathy, using GTK+ libraries and Telepathy framework, GPL

• Jitsi, a Java VoIP and Instant Messaging client with ZRTP encryption– runs on Windows, Linux and Mac OS X, LGPL

• KPhone, using Qt libraries, GPL. Runs on Linux.

• Linphone, with a core/UI separation, the GUI is using GTK+ libraries. It runs on Linux, Windows, Mac OS X and mobile phones (Android, Iphone).

• PhoneGaim, based on Pidgin[citation needed]– Does not work with new accounts after Google acquired Gizmo5.

• QuteCom, formerly known as OpenWengo, using Qt libraries, GPL

• SFLphone, with GTK+ GUI, GPL, also supports IAX2 protocol

• Telephone, Mac OS X softphone written in Cocoa/Objective-C

• Twinkle, using Qt libraries, GPL

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Free and open source mobile SIP clients

• Sipdroid for Android, GPL v3

• Linphone for Android, Blackberry, and iPhone, GPL v2

• CSipSimple for Android, GPL v3

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Tools for debugging and testing and other support

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Wireshark

• Network protocol analyzer

– With real time capture & analysis support

• Supports SIP and Diameter

• Download at:

– www.wireshark.org/

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SIPp• A SIP testing and performance

measurement tool.

• Write test scenarios to be executed against your service in a XML format

– Which requests to be sent

– Which responses to expect

– Other action to be taken

• Download at sipp.sourceforge.net/

<?xml version="1.0"?>

<!DOCTYPE scenario SYSTEM "sipp.dtd">

<scenario name=“Register with IMS">

<send retrans="900">

<![CDATA[

REGISTER sip:[field1] SIP/2.0

Via: SIP/2.0/[transport]

[local_ip]:[local_port]

To: <sip:[field0]@[field1]>

From: <sip:[field0]@[field1]>

Contact:

<sip:[field0]@[local_ip]:[local_port]>;transpor

t=[transport]

Expires: 300

Call-ID: [call_id]

CSeq: [cseq] REGISTER

Content-Length: 0

]]>

</send>

<recv response="401" auth="true"/>

</scenario>

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EventHelix IMS Call Flows

• EventStudio System Designer 5

– http://www.eventhelix.com/

• Real-time Mantra

– Detailed IMS call flow diagrams for a set of typicalfollowing scenarios

– Telecom Call Flows• Detailed call flow diagrams for major telecommunications

protocols:

– GSM, SIP, H.323, ISUP, V5.2 and IMS

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EventHelix sample IMS call flows

• Detailed IMS call flow diagrams for the following scenarios:– IMS Registration Sequence Diagrams

– IMS to IMS Call Flow

– PSTN (ISUP) to IMS Interworking Call Flow

– IMS to PSTN (ISUP) Interworking Call Flow

– SDP Codec Selection and QoS Signaling in an IMS Call

– IMS Presence Subscription and Notification Flow

– IMS Presence Resource List Subscription and Notification Flow

– IMS Conference Call

– Push-to-talk over Cellular - Pre-established Session

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Eventhelix sample IMS call flow - registration

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EIMS

Project is co-financed by European Union within European Social Fund62EIMS - 2015