Final year project in JIIT on secured voip calling.

1. Android based Encrypted IP
Voice Communication on GSM
Data Channels (GPRS) using SIP
Server

2. Android is part of the ‘build a
better phone’ process
Open Handset Alliance produces
Android
Open Handset Alliance produces
Android
Comprises handset manufacturers,
software firms, mobile operators, and
other manufactures and funding
companies
Comprises handset manufacturers,
software firms, mobile operators, and
other manufactures and funding
companies

3. Android applications are written
in Java
package com.google.android.helloactivity;
import android.app.Activity;
import android.os.Bundle;
public class HelloActivity extends Activity {
public HelloActivity() {
}
@Override
public void onCreate(Bundle icicle) {
super.onCreate(icicle);
setContentView(R.layout.hello_activity);
}
}

4. Android applications are
compiled to Dalvik bytecode
Write app in JavaWrite app in Java
Compiled in JavaCompiled in Java
Transformed to Dalvik bytecodeTransformed to Dalvik bytecode
Linux OSLinux OS
Loaded into Dalvik VMLoaded into Dalvik VM

5. The Dalvik runtime is optimised
for mobile applications
Run multiple VMs efficientlyRun multiple VMs efficiently
Each app has its own VMEach app has its own VM
Minimal memory footprintMinimal memory footprint

6. Android has many components

7. Can assume that most have
android 4.0 or above
Bruce Scharlau, University of Aberdeen, 2010
http://developer.android.com/resources/dashboard/platform-versions.html

8. Introduction
SIP is
• An Application-layer control (signaling)
protocol for creating, modifying and
terminating sessions with one or more
participants.
• Sessions include Internet multimedia
conferences, Internet telephone calls and
multimedia distribution.
• Members in a session can communicate via
multicast or via a mesh of unicast relations, or
a combination of these.
• Text based , Model similar to HTTP : uses
client-server model

9. SIP Basic Functionality
Supports 5 facets of communication:
• User location: determination of the end
system to be used for communication;
• User capabilities: determination of the media
and media parameters to be used;
• User availability: determination of the
willingness of the called party to engage in
communications;
• Call setup: "ringing", establishment of call
parameters at both called and calling party;
• Call handling: including transfer and
termination of calls.

10. SIP Functionality (cont.)
• SIP can also initiate multi-party calls
using a multipoint control unit (MCU) or
fully-meshed interconnection instead of
multicast.
• Internet telephony gateways that
connect Public Switched Telephone
Network (PSTN) parties can also use
SIP to set up calls between them.

11. Development of SIP
• SIP developed by Handley, Schulzrinne,
Schooler, and Rosenberg
- Submitted as Internet-Draft 7/97
• Assigned RFC 2543 in 3/99
• Goals: Re-use of & Maximum Interoperability
with existing protocols
• Alternative to ITU’s H.323
- H.323 used for IP Telephony since 1994
- Problems: No new services, addressing,
features
- Concerns: scalability, extensibility

12. SIP Philosophy
• Internet Standard
- IETF - http://www.ietf.org
• Reuse Internet addressing (URLs, DNS,
proxies)
- Utilizes rich Internet feature set
• Reuse HTTP coding
- Text based
• Makes no assumptions about underlying
protocol:
- TCP, UDP, X.25, frame, ATM, etc.
- Support of multicast

13. SIP Architecture
• SIP uses client/server architecture
• Elements:
– SIP User Agents (SIP Phones)
– SIP Servers (Proxy or Redirect - used to
locate SIP users or to forward messages.)
• Can be stateless or stateful
– SIP Gateways:
• To PSTN for telephony interworking
• To H.323 for IP Telephony interworking
• Client - originates message
• Server - responds to or forwards message

14. SIP Entities
• User Agents
– User Agent Client (UAC): Initiates SIP requests
– User Agent Server (UAS): Returns SIP responses
• Network Servers (diff. types may be co-located )
– Proxy: Decides next hop and forwards request, relays call
signaling , operates in a transactional manner, saves no
session state
– Redirect: Sends address of next hop back to client,
redirects callers to other servers
- Registrar: Accepts REGISTER requests from clients,
maintains users’ whereabouts at a location server

15. SIP Session Establishment and
Call Termination
From the RADVISION whitepaper on SIP

16. SIP Call Redirection
From the RADVISION whitepaper on
SIP

17. Call Proxying
From the RADVISION whitepaper on SIP

18. Instant messaging based on
SIP
• SIMPLE – IM protocol based on SIP
• SIP promises interoperability between
various IM vendors
• “Forking proxy “
• SIP has unique user tracking features.
• SIP addressing

19. Instant Messaging (Contd.)
SIP Client SIP Client
dynamic.com
columbia.edu
SIP Redirect
server
SIP proxy
foo.com
Location
service
proxy
sales.foo.com
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5
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10
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20. SIP for Internet Telephony
• Two types of phones – IP phones and
conventional analog phones.
• Uses phone numbers instead of IP
addresses
• To place a call to an IP phone, DNS is
used
• To place a call to an analog phone,
gateway
protocols like BGP are used

21. SIP Protocol Use
Henning Schulzrinne’s tutorial on SIP

22. Where’s SIP
Application
Transport
Network
Ethernet
IP
TCP UDP
RTSP SIP
SDP codecs
RTP DNS(SRV)

23. Conclusions
• Through this application, the users will be
able to communicate with each other
securely using GPRS
• We shall use strongest encryption
technique called ZRTP which would be
layered on SIP protocol for an effective