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Mobile SIP COMMUNICATOR Date: 24 March 2012

Mobile SIP (Session Initiation Protocol) Communicator

Group Members:-PRASAD PURNAYEJUGAL PORWALAJINKYA PHALKEMAHESH TEKADE

Seminar Guide:-Prof. M.V.Marathe


Agenda

Introduction to SIP(Session Initiation Protocol)

Components & Elements of Session Initiation Protocol

SIP Functioning

SIP Communicator –Mobile Version Implementation


>Traditionaltelephony

>Internet telephony

internet telephony


VoIP

H.323

• By ITU_T(International Telecommunication Union_ Telecommunication Standards)

• For Multimedia Application on telephony

H.323 drawbacks

• Requires significant work to scale the apps with protocol

• First Establish the connection and then negotiates the Capabilities and Features


HISTORY OF SIP

REUSE OF COMMON ELEMENTS

HTTP-

DNS -

SMTP-

404

SRV

MIME

“The Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol for creating, modifying and terminating sessions with one or more participants.”

Can be used for voice, video, etc..

Follows on HTTP & SMTP


What is SIP

Application

Transport

Network

Physical/Data Link Ethernet

IP

TCP UDP

RTSP SIP

SDP codecs

RTP DNS(SRV)


WHERE IS SIP LOCATED?

H.323 Standard

Jabber (XMPP)

SCCP(Skinny Client Control Protocol By Cisco )

SIP (Session Initiation Protocol)

Inter-Asterisk Exchange Protocol by Asterisk (IAX2)


Some of the Protocols that Uses VoIP

SIP (Session Initiation Protocol)H.323 Standard


•Addressing mode- URI(Uniform Resource Indicators)

•Call establishment Methods.

•Network administration level.

•mobility

Jabber (XMPP)

XML (extensible markup language) based protocol.

Originally its scope was only instant messaging.

Uses RTP for data transmission

Adopted of XMPP by Google for use in its Google Talk service


SCCP(Skinny Client Control Protocol By Cisco )

protocol used by Cisco in its CallManager server and associated phone terminals.

It is a proprietary protocol so publicinformation is scarce.


Inter-Asterisk Exchange Protocol by Asterisk (IAX2)

IAX protocol is that it combines both signaling and voice data into the same data stream.

The best known VoIP related Free Software server is Asterisk.


Locating Users and Resolving Their SIP address To IP Address


SIP FEATURES

Manage the Setup And tear down the calls for ALL USERS IN THE SESSION


SIP FEATURES

Negotiation Capabilities among ALL SESSION PARTICIPANTS


SIP FEATURES

Changing Session parameters DURING THE CALL


SIP FEATURES



SIP Functioning




User Agents(clients & Server)

Proxy Server

Registrar Redirect Server

SIP


SIP NETWORK ELEMENTS

User Agents (Clients & Server)

Proxy server

Registrar

Redirect server

NETWORK ELEMENTS



• SIP enabled device is called User Agent

• Takes direction or input from a user and acts as an agent on their behalf to set up and tear down media sessions with other user agents

• A UA should advertise its capabilities and features in any request it sends. This allows other UAs to learn them without having to make an explicit capabilities query

USER AGENTS


1)Presence Agent

• It authenticates a subscription request.

• If the authentication passes, it establishes a dialog and sends the notifications over that dialog.

CATEGORIES OF SIP USER AGENTS


2)Back to Back User Agent(B2BUA)

• Receives a SIP request, then reformulates the request and sends it out as a new request.

• B2BUA device can be used to implement an anonymizer service in which two SIP UAs can communicate without either party learning the other party’s URI, IP address, or other information

• However it reduces reliability of SIP Sessions over Internet because it breaks end-to-end nature of an Internet protocol such as SIP

CATEGORIES OF SIP USER AGENTS


• SIP servers are applications that accept SIP requests and respond to them

• servers provide services and features to user agents, they must support both TCP and UDP for transport

SIP Servers

Mobile SIP (Session Initiation Protocol) Communicator1)Proxy Servers

A SIP proxy server receives a SIP request from a user agent or another proxy and acts on behalf of the user agent in forwarding or responding to the request

A proxy is not a B2BUA since it is only allowed to modify requests and responses according to strict rules set out in RFC 3261.

Some further sub types of Proxy Servers

a)Stateful Proxy Serverb)Stateless Proxy Server


2)Redirect ServersA redirect is a type of SIP Server that responds to, but

does not forward, requestsIt comes into picture when the callee is outside the

domain of caller.


Registrar Server

•A SIP registrar server was introduced in the example of Figure 3.3. A registrar server, also known as a registration server, accepts SIP REGISTER requests; all other requests receive a 501 Not Implemented response

• The contact information from the request is then made available to other SIP servers within the same admin- istrative domain, such as proxies and redirect servers


Star Point:FOR SECURED REGISTRATION OF THE USER,SIP USES SOME OF THE FOLLOWING PROTOCOLS:1)IPSec 2)TLS 3)DNSSec

Registrar Server


TCP

(Transmission Control Protocol)

RTP

(Real-time Transmission Protocol)

UDP

(User Datagram Protocol)

RTCP

(RTP Control Protocol)

SIP

SIP Components

RTP(Real-time Transport Protocol)


•Real-Time Transport Protocol was developed to enable the transport of real- time datagrams containing voice, video, or other information over IP.

• Both H.323 and SIP use RTP for media transport, making it the most common standard for Internet communications.



RTP allows for the detection of some of the impairments introduced by an IP network, such as:

• Packet loss;• Variable transport delay;• Out of sequence packet arrival; • Asymmetric routing.



How RTP fits into the common media processing steps?1. Coding.2. Packetization3. Transport.4. Depacketization.5. Buffering.6. Decoding.7. Playback.



In terms of media quality, the two most important factors area)Packet Loss Rateb)End to End Latency

1) RTP media sessions are unidirectional—they define how media is sent from the media source to the media sink.

2) In order to avoid any third party from interrupting the session we use Secure RTP(SRTP)

RTCP(RTP Control Protocol)


It allows participants in an RTP session to send each other quality reports and statistics, and exchange some basic identity information.The five types of RTCP packets are shown as follows :1. SENDER REPORT(SR)2.RECEIVER REPORT(RR)3.SOURCE DESCRIPTION(SDES)4.BYE(BYE)5.APPLICATION SPECIFIC(APP)

RTCP(RTP Control Protocol)


The use of reports allows feedback on the quality of the connection including information such as:• Number of packets sent and received;• Number of packets lost;• Packet jitter.

1)Designed to scale for very large conferences2)RTCP Extended Reports

a).RTCP extended reports defines seven additional report blocks.They were defined due to limitation of the basic SR and RR.

b.)In addition, RTCP-XR defines a way to estimate actual voice call quality and exchange this information.



SIP Functioning



Request Response


INVITE

REGISTER

BYE

ACK

CANCEL

OPTIONS

SIP REQUEST MESSAGES


INVITE

used to establish media sessions between user agents.

Has a message body containing the media information of the caller.


Mandatory Header Fields in an INVITE

ViaToFromCall-IDCSeqContactMax-Forwards


INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP lab.high-voltage.org:5060;branch=z9hG4bKfw19bMax-Forwards: 70To: G. Marconi <sip:[email protected]>From: Nikola Tesla <sip:[email protected]>;tag=76341Call-ID: j2qu348ek2328wsCSeq: 1 INVITESubject: About That Power Outage...Contact: <sip:[email protected]>


REGISTER

The REGISTER method is used by a UA to notify a SIP network of its currentContactA REGISTER request may contain a message body, although its use is not defi ned in the standard.


Mandatory Header Fields in a REGISTER

ViaToFromCall-IDCseqMax-Forwards


REGISTER sip:registrar.athens.gr SIP/2.0Via: SIP/2.0/UDP 01.202.203.204:5060;branch=z9hG4bK313Max-Forwards:70To: sip:[email protected]: <sip:[email protected]>;tag=543131Call-ID: 48erl8132409wqerCSeq: 1 REGISTERContact: sip:[email protected]: mailto:[email protected]: 0


BYE

The BYE method is used to terminate an established media session. A BYE is sent only by UAs participating in the session, never by proxies or other third parties.A BYE cannot be used to cancel pending INVITEs because it will not be forked like an INVITE and may not reach the same set of UAs as the INVITE.


Mandatory Header Fields in BYE



BYE sip:[email protected] SIP/2.0Via: SIP/2.0/TCPport443.hotmail.com:54212;branch=z9hG4bK312bcMax-Forwards:70To: <sip:[email protected]>;tag=63124From: <sip:[email protected]>;tag=9341123Call-ID: 34283291273CSeq: 47 BYEContent-Length: 0


ACK

The ACK method is used to acknowledge final responses to INVITE requests.Final responses to all other requests are never acknowledged.


Mandatory Header Fields in an ACK



ACK sip:[email protected] SIP/2.0Via: SIP/2.0/TCP 128.5.2.1:5060;branch=z9hG4bK1834Max-Forwards:70To: Marquis de Laplace <sip:[email protected]>;tag=90210From: Nathaniel Bowditch <sip:[email protected]>;tag=887865Call-ID: 152-45-32-N-32-23-47-W


CANCEL

used to terminate pending INVITEs or call attemptsCan be generated by both User Agents Or Proxy Servers


Mandatory Header Fields in CANCEL



CANCEL sip:[email protected] SIP/2.0Via: SIP/2.0/UDP 10.downing.gb:5060;branch=z9hG4bK3134134Max-Forwards:70To: Isaac Newton <sip:[email protected]>From: Rene Descartes <sip:[email protected]>;tag=034323Call-ID: 23d8e0e4e2e505329299e288bbd4155aCSeq: 32156 CANCELContent-Length: 0


OPTIONS

The OPTIONS method is used to query a user agent or server about its capabilities and discover its current availability.May Not contain Message body


Mandatory Header Fields in an OPTION



OPTIONS sip:[email protected] SIP/2.0Via: SIP/2.0/UDP cavendish.kings.cambridge.edu.uk;branch=z9hG4bK1834Max-Forwards:70To: <sip:[email protected]>From: J.C. Maxwell <sip:[email protected]>;tag=34Call-ID: 747469e729acd305CSeq: 29 OPTIONSContent-Length: 0


1XX Provisional 100 Trying

2XX Successful 200 OK

3XX Redirection 302 Moved temporarily

4XX Client Error 404 Not Found

5XX Server Error 504 Server Time-out

6XX Global Failure 603 Decline

SIP RESPONSE MESSAGES


SIP Flows - Basic

ACK

200 - OK

INVITE: sip:18.18.2.4“Calls”

18.18.2.4

180 - Ringing Rings

200 - OK Answers

BYEHangs up

RTPTalking Talking

User A

User B


200 - OK

REGISTER: sip:[email protected]

401 - Unauthorized

REGISTER: (add credentials)


Registrarclient

INVITE: sip:[email protected]“Calls” dbaron

@MIT.EDUINVITE: sip:[email protected]

180 - Ringing

Rings180 - Ringing

200 - OK Answers

200 - OK

ACK

BYEHangs up

200 - OK

User A

User B

MIT.E

Proxy

Talking TalkingRTP


INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP 100.101.102.103:5060;branch=z9hG4bKmp17aMax-Forwards: 70To: Heisenberg <sip:[email protected]>From: E. Schroedinger <sip:[email protected]>;tag=42

INVITE sip:[email protected]: SIP/2.0/UDP proxy.munich.de:5060;branch=z9hG4bK83842.1Via: SIP/2.0/UDP 100.101.102.103:5060;branch=z9hG4bKmp17aMax-Forwards: 69To: Heisenberg <sip:[email protected]>From: E. Schroedinger <sip:[email protected]>;tag=42

INVITE: sip:[email protected]“Calls” joe @MIT.EDU

INVITE: sip:[email protected]

100 - Trying

ACKACK

User A MIT.EDU

Proxy

30161Gateway

180 - Ringing

180 - Ringing

Rings

200 - OK

200 - OK

Answers

BYEHangs up

BYE

200 - OK

200 - OK

Talking TalkingRTP





SIP Functioning


Idea of Mobile Implementation

Platform of working J2ME (on ANDROID)

Requirement Specification in terms of Network Service provider

Different Modules used in application

Stepwise Working of communicator

Feasibility of the Project



Classes

SipAudioCall

SipAudioCall.Listener

SipErrorCode

SipManager

SipProfile

SipProfile.Builder

SipSession

SipSession.Listener

SipSession.State

SipRegistrationListener


Classes used

SIP MANAGER

• register()• unregister()• isRegistered() • isVoipSupported()• makeAudioCall()• makeVideoCall()• newInstance()• open()• takeAudioCall()• takeVideoCall()


PROFILE BUILDER

Classes used

•setAuthUserName()•setAutoRegistration()•setDisplayName()•setPassword()•setPort()•setProfileName()•setProtocol()


PEER PROFILE

Classes used

•getAuthUserName()•getDisplayName()•getPassword()•getSipDomain()•getPort()•getProfileName()•getProtocol()•getProxyAddress()•getSipDomain()


AUDIO CALL

Classes used

•getLocalProfile()•getPeerProfile()•answerCall()•attachCall()•holdCall()•continueCall()•endCall()•startAudio()•toggleMute()


AUDIO Call Listener

Classes used

•onCallBusy()•onCallEnded()•onCallEstablished()•onCallHeld()•onCalling()•onChanged()•onError()•onReadyToCall()•onRinging()•onRingingBack()


ERROR CODE

Classes used

•CROSS_DOMAIN_AUTHENTICATION()•DATA_CONNECTION_LOST()•INVALID_CREDENTIALS()•INVALID_REMOTE_URI()•IN_PROGRESS()•NO_ERROR()

•PEER_NOT_REACHABLE()•SERVER_ERROR()•SERVER_UNREACHABLE()•SOCKET_ERROR()•TIME_OUT()•TRANSACTION_TERMINATED()


SESSION

Classes used

•answerCall()•changeCall()•endCall()•getCallId()•getLocalIp()•getLocalProfile()•getPeerProfile()

•getState()•isInCall()•makeCall()•Register()•setListener()•Unregister()


SESSION STATE

Classes used

•REGISTERING()•INCOMING_CALL()•INCOMING_CALL_ANSWERING()•IN_CALL()•NOT_DEFINED()•OUTGOING_CALL()•OUTGOING_CALL_CANCELING()•OUTGOING_CALL_RING_BACK()•DEREGISTERING()

Register to SIP server

Profile Exists

Request For New Session

Ask For Peer Profile

Establish Connection

Session Initiated

End Session

YES

Create New Profile

NO

Manager

Register()

isRegistered()

newInstance()


Profile Exists




Session Initiated

End Session

YES

Create New Profile

NO

Profile Builder

setAutoRegistration()

setDisplayName()

setPassword()

setPort()

setProfileName()

setProtocol()


Profile Exists




Session Initiated

End Session

YES

Create New Profile

NO

Session

getCallId

getLocalIp

getLocalProfile

changeCall


Profile Exists




Session Initiated

End Session

YES

Create New Profile

NO

Profile

getAuthUserName()

getDisplayName()

getPassword()

getPort()

getProfileName()

getProtocol()

getProxyAddress()

getSipDomain()


Profile Exists




Session Initiated

End Session

YES

Create New Profile

NO

AudioanswerCall()

attachCall()

close()

continueCall()

endCall()

holdCall()

isInCall()

isMuted()

isOnHold()

makeCall()

setSpeakerMode()

startAudio()

toggleMute


Profile Exists




Session Initiated

End Session

YES

Create New Profile

NO

Manager

Unregister()

Close()


Advantages

Enables Data and Voice convergence

Mobility

Enhanced audio quality

Integrated Presence

Simplicity & Extensibility

Can run over reliable & unreliable transport protocols


Conclusion

1. SIP, though not used extensively in day to day life, is a evolving protocol. Which in future can revolutionalise the internet telephony system.

2. The rapid conversion to SIP is a strong step towards inter-operability.i.e various vendors will inter-operate for the better functioning.

3. Numerous application level features and capabilities are developed to advance mobility and productivity fot business and their end users.

4. SIP based communications deliver a suite of solutions that can significantly enhance users’ communication options and productivity.


References

1. H. Schulzrinne, “Re-engineering the telephone system,” Proc. of IEEE Singapore Intl. Conf. on Networks (SICON) 1997: The Next Millennium, Singapore, Apr. 14–17, 1997, pp. 261–275.2. H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, “RTP: a transport protocol for real-time applications, Request for Comments (Proposed Standard) 1889,” IETF, Jan. 1996.3. B. Braden, L. Zhang, S. Berson, S. Herzog, and S. Jamin, “Resource ReSerVation protocol (RSVP)—version 1 functional specification, Request for Comments (Proposed Standard) 2205,” IETF, Oct. 1997.4. B. Braden and L. Zhang, “Resource ReSerVation protocol (RSVP) version 1 message processing rules, Request for Comments (Proposed Standard) 2209,” IETF, Oct. 1997.

Thank you !!

Sip

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