Study Mechanism for Providing Emergency Services using VoIP/IP Network (Match 2014

Department of Computer Engineering Oklahoma Christian University

Abstract —The current communication infrastructure of 9-1-1 is based on analog

technology that doesn’t support many advanced telecommunication features

(texting, emailing picture…) that most end users are expecting to be part of emergency

communication system.To remedy these problem emergency services have to be provided using

IP networks. Providing Voice over Internet Protocol (VoIP) based Emergency services is

not only a success of VoIP but it will also give opportunities to enhance the existing

emergency call processing infrastructure. The intent of this paper is to study mechanism

that can be used to provide emergency services using IP Networks; we will discuss

how VoIP can be used to provide Emergency services, how to identify emergency calls,

determining the caller’s locations and routing him/her to the appropriate Public

Safety Answering Points will be addressed and then finally we will discuss how Internet can

be used to alert the public about the pending or current emergency situations.The solution

that we propose is based on Session Initiation Protocol (SIP).

Index Terms —Emergency Network, SIP/VoIP, emergency alerting,emergency communication

I.INTRODUCTION

V oice over Internet Protocol (VoIP) is a technology that allows communication IP

networks instead of traditional analog system. VoIP is becoming a predominant

Desire Aheza is with the Computer Engineering Department,Oklahoma Christian University,Edmond,OK,(e-

mail:[email protected]).

technology in the world of telecommunication systems. As many people around the world are adopting VoIP due to its attractive service

with advanced features (video calls…)and costsaving.

Everyone around the world expect to be able to reach 9-1-1 or 1-1-2 in Europe in order to

obtain emergency assistance from their public safety agencies i.e.First Responders

(fire fighter,medical,law enforcement).With the growing popularity of VoIP services,3G/4G

wireless phones, and other advanced communications technologies, customers

subscribed to these technologies expect that they can use them to reach 911 services,just

as they can by using ordinary public landline communication system to reach 9-1-1. Most of

the subscribers are unaware of the current limitations of the 9-1-1 network

infrastructure with respect to some wireless technology or internet based technology.

Sometime they got disappointed due to the fact that the technology that they use most

the time can’t help them to reach 9-1-1. This has been enhanced when a student at Virginia

Tech, have tried to reach 9-1-1 using text message and disappointed when he found that

text message are not supported to his local PSAP.

With this growth in use of VoIP,it is evident that calls to 9-1-1 Public Safety Answering Point will be

originated from these VoIP users. The high expectations of the public that they can

reach the 9-1-1 systems using any communication device, testifies the need of

EMERGENCY SERVICES OVER IP NETWORK.This will enhance the current emergency systems as IP

networks allow high speed redundant links,

Study Mechanism for Providing Emergency Services using VoIP/IP

Network (Match 2014)

Desire AHEZA,Department of Computer Engineering Oklahoma Christian University, Edmond,OK

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Failover, backup systems, Standardized protocols, resilient and reliable networks,

faster call setup, faster response and lower cost.

In this project we study a mechanism that can be adopted to achieve the enhancement of the

emergency network. The mechanisms will be based on an architecture that can provide

emergency service in VoIP network based on Session Initiation Protocol (SIP) which is a

signaling communications protocol, used for setting, maintaining and terminating

multimedia communication sessions such as voice and video calls over Internet Protocol

(IP)networks 1. The following section presents an

introduction to the current emergency network and its deficiencies.Section

Current 9-1-1 network infrastructure

The current systems for 9-1-1 calls today operate almost exclusively on analog

communication technology, using an architecture of circuits switched networks

developed when the American Telephone & Telegraph Company was a regulated monopoly providing most of the nation’s telephone

service2 .Systems for 9-1-1,nowadays is unable to accommodate the latest advances or

features in telecommunications technology, the 9-1-1 systems are increasingly outdated,

costly to maintain, and in danger of failure with many single point of failure entities in

the current emergency communication network3. The old (1970s)network architecture of 9-1-1 is

still in use in some parts of emergency services in the U.S. nowadays. The kind of

architecture is presented in figure 1, this was simply the switching in the local

telephone company central office (CO)that was interpreting the dialed sequence of 9-1-1 then

route the call to the local single Public Safety Answering Point see Figure 1. No

additional information is transmitted along with the call.

Figure1:Basic 9-1-1 network architecture

This architecture suffers 3 significantlimitations1:

There is no automatic mechanism to deliver the telephone number of the caller this means that no callback number available to the calltaker

(dispatcher) that he/she can use in case the call has been disconnected.

Secondly the architecture in Figure 1 can’t provide to the calltaker the information related to the

LOCATION of the calling party summoning emergency assistance.

The last constraint is that routing calls to wrong PSAPs might occur. This is due to the

fact that geographic area served by a certain telephone carrier; their CO might not match

the relevant political boundaries to the proper PSAP.

In this architecture it is relevant that time is wasted every time in emergency events

because no location information is identified.

Over the time, progresses in technology were taken to overcome all of these 3 limitations.

The 9-1-1 system has evolved to Enhanced 9-1-1 i.e.E9-1-1 which is an emergency system that can handle new forms of call delivery (VoIP

calls and call from Wireless mobile),although the core of the E9-1-1 was still is based on the analog technology.

The adoption of the Automatic Number Identification (“ANI”)has solved the issue of

the CALL BACK NUMBER on the caller taker post. The caller number has to be captured by the CO

switch and relayed to PSAP along with the calls summoning the emergency in order to

provide the calltaker the CALLBACK NUMBER1 . The lack of the location information in the

legacy 9-1-1 was overcome by creating and using a dedicated database for emergency service1.

This is a time Local Exchange Carrier (LEC)

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that was in use for the billing purpose, its role was to associate particular phone lines

and numbers with corresponding subscribers identities (Address and Name). Information

queued in the LEC were used to create a specialized database in 9-1-1 known as

Automatic Location Identification (ALI)with major takes of associating telephone numbers

and the names and the address information.But still constraints associated to the analog

platform were still present as the PSAPs were using SEPARATE LINKs to access the ALI

databases.

To overcome these limitations associated to the routing of emergency calls to the proper

PSAP, additional equipment has been introduced” Selective Routing Database”

(“SRDB”). In this architecture, emergency calls (9-1-1) originating from a number of COs

are aggregated at the SRDB and then delivered over a group of trunks to the proper PSAP1 .Note that also a Master Street and Address Guide

(MSAG)has been introduced in order to create the logical association between the proper

PSAP and the calling party’s number. This is was a major advancement in the 9-1-1; MSAG

associates streets addresses and telephone numbers to their corresponding (particular)

Public Safety Answering Point and provides details about the different emergency

service agencies that can respond to that particular location.

Figure 2: E9-1-1 network architecture that support

VoIP and Wireless

The architecture presented in figure 2 is the current E9-1-1 network, this architecture

support VoIP subscribers and wireless mobile subscribers.

The work to provide emergency service to wireless subscribers has been carried in two

phases; under Phase 1, wireless carriers was supposed to provide to the PSAPs the mobile

call back number and the location of the Base Station Transceiver (BTS)i.e.the antenna that receive the call to 9-1-1. Under phase 2 the

latitude and longitude coordinates (estimated coordinates)of the caller’s

wireless device are provided in additional to the information provided in phase1.

In mobile wireless network,Mobile Switching Centers (MSC) equivalent to CO in PSTN have

been introduced; when MSC receives an emergency call requesting connection to 911 services, a processor associated with that

switch has to know the antenna sector where the call to 9-1-1 is originated and then

selects one of the available (not in use) pseudo ANIs for the associated set to that

antenna for routing purpose. Then the Mobile Switching Center forwards the emergency call

along the pseudo ANIs to the E9-1-1 CO, simultaneously the MSC also send antenna

location details to the ALI database such that the Public Safety Answering Point can

access it for further processing. In this architecture PSAP receives pseudo ANIs and 9-

1-1 call from the CO, using these information it can queries the ALI database then receives in return the CALLBACK NUMBER with the

information about the BTS SECTOR of the CALLER’s Mobile device1.

Phase 2 has introduced new equipment in the network i.e. the Position Determination

Equipment (PDE), this one approximates (estimation) the location of the subscriber

using wireless mobile device to place calls to 9-1-1.This is performed both at the start of

the call or during the progress of the call depending when information is needed1 .

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To introduce VoIP devices in 9-1-1, VoIP providers have obliged their customers to

manually provide their address over the Internet and then the caller location

information were directly forwarded to the proper 9-1-1 databases. With this

architecture presented in figure 2, an emergency VoIP call to 9-1-1 is passed, along

with the ANI details,through “VoIP gateways” that are used to convert digital packet into

analog format and then pass it to the E9-1-1 CO and, then passed it to the appropriate PSAP. Then the Customer Premise Equipment at the PSAP uses the connection to the ALI database

to retrieve and display the address, subscriber name, callback number, and

emergency services details corresponding to the calling party’s number.

However the infrastructure presented in Figure 2 presents some constraints

associated with the use of analog technology1:

i. Limitations of Exploiting and/or Serving Current Technology

Subscriber Devices

The technology of 9-1-1 that has been established in 1968was developed for users of

ordinary telephone handsets. Today’s the public has switched to VoIP and wireless

devices that use different technologies (digital) comparing to ordinary telephone handsets.These digital devices have built in

processors and memory chips that are capable of creating and handling not only voice

messages but also text messages, data, video, and image signals and multimedia information

as well. Protocols have been developed to allow these devices to communicate using packet switched networks.These networks are

capable of transporting multimedia signals from these digital devices.

Current technology implemented in PSAPS is not capable of exploiting or serving these

modern digital devices, if a subscriber sent to 9-1-1 for example a picture of a car leaving

a crime scene the PSAP will not have the technology to treat the kind of message

because it has been designed with voice capability only. We need a new technology to

overcome this limitation

ii. Constraints associated with analog circuit switching

and selective routers

The current 9-1-1 network can only handle voice traffic/data in a world of digital technology.

Most of the equipment used in the current 9-1-1 network architecture are optimized to

support only voice communication1 ,and hence there is limited bandwidth available for data

communication. The use of Selective Routers (SR) in the

current architecture of 9-1-1 also create single point of failure; let’s say if the SR fails, all calls to 9-1-1 routed via the CO

connected to that SR will not reach to their destination.To correct this issue;redundant

links to two different SRs located in different locations are required. Note that

also the same problem of one point of failure exists as the PSAPs have no back up

infrastructures. The current architecture based on analog technology cannot allow PSAPs

to back up each other in case of high volume calls at one PSAPs,call or any data exchange can’t be transferred between PSAPs and this

is due to the use of old signaling protocols.

This limit the sharing of foreign language speaking call-centers or some specialized databases or software.

iii. Limitations due to In-band Multi- frequency signaling (CAMA)

The inBand analog Multi-Frequency (MF) signaling that is used to carry the ANI

information between the PSAP and the E9-1-1 CO is an outdated form of signaling; it is slow

(slow speed this means that the call setup time is delayed;the caller can think that has

not gone through and hangs up the call)and has

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limited capacities vis-a-via to modern form of signaling (MF signals are transmitted in

analog form so they can face more errors).

With MF signaling there is no way to carry emergency call along with location and the

call back number 1 , rather the PSAPs request the necessary information to the ALI database

using a separate link thus an additional delay in the delivery of details (content)

associated with the call. All limitations between the 9-1-1 Control

Office and the PSAP are tighten to the Centralized Automatic Message Accounting

(CAMA) signaling used between these entities as it relays to analogy technology1 .

iv. Limitations due to the fixed point to point Data links

The current E9-1-1 network architecture includes a Point-to-Point (P2P) link between

the ALI database and the PSAP.According to the current technology it is unusual to find a P2P

links over low speed analogy facilities; these low speed links slow down the

connection and the time between when a query is initiated and the ALI information is

delivered is increased.Note that the lack of broadband communication links between PSAPs

limits the ability of calltakers to automatically share by transferring data

that has been collected from the calling party along with the call when it is

transferred from one PSAP to others1 . This architecture isolates PSAPs as they can’t

share expensive or specialized personnel and information between them.

Public Expectations

Public expectations for timely and accurate response to 9-1-1 calls are based on the high-

tech features presented in most of their communication devices,but not on the reality

of the current technology (analogy transmission) that the 9-1-1 system uses. For

example, analog systems cannot allow the transmission of video, picture or text

messages. Calls setup are delayed or dropped during the communication progress.There are losses of information related to the location

of the caller because the information are digital and they cannot be transmitted by

means of the current switching technology or understood (treated) by the computers at the

PSAPs centers3 . A report of the National Emergency Number Association (NENA)summarize

that the 9-1-1 network infrastructure has not kept pace with the advance in technology and

now is of the critical systems that need to bemoderated4 . The current emergency network

need to be enhanced such that it can provide the QoS that approaches the expectations and

needs of its consumers and this refers to the investments in new technologies “IP

telephony/VoIP”.

The study we conducted on the current E9-1-1 network shows that a new network

infrastructure is needed in order to provide emergency service to the public

independently of their communication technologies.

IP networks are based on the same type of network infrastructures as the one

businesses and consumers use to access to the Internet and share capacities with other

subscribers. Broadband connections built to hospitals,libraries,or campuses have also to

reach 9-1-1 PSAP/call center.

The next section introduces IP networks and the Session Initiation Protocol (SIP)which is

a signaling communications protocol,used for setting, maintaining and terminating

multimedia communication sessions such as voice and video calls over Internet Protocol

(IP)networks.

Brief VoIP Standards and Protocols Internet Telephony/VoIP is the communication

over IP networks; it allows the transmission of voice and video, voice-messaging

application and/or facsimile via Internet rather than the PSTN.The basic steps involved

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in Internet based communication is the conversion of the analog signal to digital

format, compression and translation of the signal into IP packets that can be

transmitted over the Internet;the process is reversed at the receiving end.

Signaling protocols are used to set up and tear down calls and carry information

required to locate users.

To make VoIP popular;products from different vendors need to operate with each other to

grant the interoperability, to achieve this task different standards has been developed

and the most common standards for VoIP are the Session Initiation Protocol (SIP) and H3235.

The following sections present a description of one of these standards “SIP”. Figure 3

displays the setup of a VoIP system and how audio signals are processed by it.

Figure 3 Treatment of audio signal in a VoIPsystem 5

The digitized voice from the sending speaker is encoded by a voice encoder,and then packets

are sent using a protocol stack. When the speaker is silent, the Voice Activity

Detection has to recognize that and then generate a packet without payload (sometime

no packet will be generated).Then packets are sent over the network e.g. IP LAN. When the

packet reaches the receiver side it is processed by the protocol stack. Packets

losses and transmission errors are detected and corrected. Packets might not arrive in a

constant flow;so they have to be collected in a buffer (called jitter buffer) that adjusts

time differences and the sequence (arrival) order. Finally the decoder decodes the

packets and outputs them via the sound system5.

There are a number of standards used for signaling,speech coding and the transport of

the voice packets over IP. This modularization makes VoIP flexible and the

standards interchangeable when new requirements and applications emerge. The

following sections introduce the widely used standards (protocols)in VoIP.

Standards and Protocols

IP telephony requires some protocols for establishing and controlling the connection,

which is signaling the connection. Real Time Transfer (RTP) Protocol is used to send

multimedia data packets with session setup via signaling protocol i.e.SIP.

RTP protocol

The RTP protocol is a real time protocol which is oriented to the transmission of

information in real time such as voice or video. This protocol is a user session

protocol which relies on User Datagram Protocol (UDP),making use of the checksum and

multiplexing services to allow programs which make transmission of this kind of data

handling the real time unicast or multicast transmissions. RTP does not guarantee real

time delivery of multimedia data. To achieve real time transmissions; RTP uses the Real Time Control Protocol (RTCP);which provides a

feedback about some control information.With this,it is possible to monitor the quality of

the transmission and also possible to diagnose possible network problems.

RTP is composed of 4 main fields;below each isdescribed6:

RTP Payload type:this field indicates codecs

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and the specific media encoding to use.Codecs convey the type of multimedia data (video or voice) and information about how it must be

encoded or decoded.It is important to say that changing codecs will be necessary,if we have

to adapt to the variations in bandwidth, frames indications;which mark the limitation

(start&end)of each frame.

Sequence number (SN ): packet reassembling, detection of packet losses and duplication of

packets are performed at the receiver based on SN.

Time Stamp this is one is used to rebuild the timing of the original video&audio. The

receiver uses the time stamps in order to determine the variation in time of arriving

packets.To make RTP transmission possible;at the receiver, each packet is compared with time stamp to be made.

Source ID : this is utilized to differentiate multiple incoming streams. In this paper we

use RTP as voice transmission protocol to send real time packets.

To place calls over IP network, VoIP uses 2 protocols; call-setup protocol(s) and voice

transmission protocols. Protocols involved in call-setup are known as VoIP signaling

protocol(s) i.e. (SIP, H.323 and IAX (Inter Asterisk eXchange)) are most common choices.

Next paragraph introduces SIP.

Session Initiation Protocol (SIP)

SIP is an Internet Engineering Task Force (IETF)defined signaling protocol,widely used

for controlling multimedia communication sessions such as voice and video calls over Internet Protocol. This protocol is used for

creating, modifying and terminating communication (multimedia) session between

subscribers. Sessions are considered to be communication states kept between the

sending party & the receiving party during the interactions (IP calls, distribution of

multimedia etc).The modification can involve changing addresses or ports, inviting more

participants, and adding or deleting media streams. SIP clients typically use TCP

(Transmission Control Protocol)or UDP on port numbers 5060 and/or 5061 to connect to SIP servers and other SIP endpoints.Port 5060 is

commonly used for non-encrypted signaling traffic whereas port 5061 is typically used

for encrypted traffic. The following paragraphs, explains the functional and

characteristics features of SIP:

Distributed Architecture of SIP

As shown in Figure 4 the SIP protocol defines a collection of entities that take part on a SIP

communication, which are, User Agent, Registrar and Redirect Servers, Location

Servers,Proxy Servers.

Figure 4 SIP Distributed Architecture

Location Server (LS) One of SIP entities (servers) used by the

redirect and proxy servers to obtain data about the calling’s party possible location.

By using registration process this server stores the current locations of the

subscribers.

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Figure 5 Communication example of RegistrarServer 7

Redirect Server

Those servers are UA servers that accept and receive SIP requests.It checks requests from

the LS databases and produces list of current position of the subscriber and it uses some of

3xx responses to send back the request to the sender. When the UA client receives the list

(showing the exact destination position) it sends requests directly to appropriate

destination.See figure 6 for more details.

Figure 6 Redirect Server operational

behaviors 7

SIP network entities exchanges between them, SIP transactions are composed of sequence of SIP messages (request&responses) in order to

communicate.These transactions are utilized in many steps in order to complete the call

process7.

SIP messages

SIP is a text-based protocol with syntax similar to that of Hyper Text Transfer Protocol (HTTP).There are two different types

of SIP messages: requests and responses. The first line of a request has a method ,defining

the nature of the request, and a Request-URI

(Uniform Resource Indictor),indicating where the request should be sent.The first line of a

response has a response code 6

We need method in field of the request line because it is used to determine the function

of request; there are 6 types of SIP methods, these methods are explained below;

REGISTER:Used by a UA to indicate its current IP address and the URIs (Uniform Resource

Locators) for which it would like to receive calls. INVITE: Used to establish a media

session between user agents. ACK:Confirms reliable message exchanges

CANCEL:Terminates a pending request (this is activated when the calling party want to

refuse the call invitations from Calledparty).

BYE messages are utilized to close communication sessions.

OPTIONS: Requests information about the capabilities of a caller,without setting up a

call (Information discovery:about extension, supported method,type of contents,codecs …).

It allows that discovery without “ringing” the called party6.

SIP Responses

When a UA receives requests it has to reply using response methods. These responses

matches requests messages, except to the first line; the response’s first line is

composed of the protocol version number (SIP/2.0),reason phrase and reply code. The

reply code is integer number from 100 to 699 that indicates the response type.

There are 6 classes of responses:1xx : called provisional or informational

responses. They indicate that request has been received and is being processed; we can

say for example 181 to mean that the call is being forwarded,100 trying to connect and 180

Ringing2xx : This type shows that processing of calls

is successfully and accepted. This type of message (200 OK) is sent back to indicate that

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call is successful.3xx: i.e.redirect response,gives the current

location details of the calling party.4xx : These responses indicate at the sender

side that there is problem (bad syntax or negative responses from the receiver) and it

cannot be treated at the proxy.5xx : Servers’ failures response that means

the request is valid and the servers can’t process the request.

6xx : Global Failure responses.No server that can process requests and they also show that

declines response from UA when it doesn’t want to participate in the session.

Architecture of the 9-1-1 network over IPNetwork

IP telephony carries multimedia as packets over IP networks i.e.Internet. These packets

are transmitted using RTP protocol with sessions established using signaling protocols for example SIP or H.323. In this

project we will focus on the use of SIP as it the most widely used signaling protocol in

VoIP. SIP subscribers are identified either by an SIP URIs such as

sip:[email protected] or by ordinary E164 telephone numbers.

In general handling emergency calls can be done in 4 steps that might be executed in

sequences for each emergency call (figure 7).

Figure 7 steps in emergency call handling

It the first step the user agent (UA) client and the proxy server determine if the

multimedia message (example:call)is destined to 9-1-1 or not . It the call is identified as an

emergency call then the second step determine the calling party’s location,and forward the

location information along with setup messages (INVITE). The next step is to

determine the correct PSAP based on the location information from step 2 and the proxy

will route call to that PSAP. The last step presents the emergency call to the call-taker

at the PSAP. The information within the call setup can be utilized by the dispatcher at the

PSAP to handle the emergency call or plan for the type of the rescue to send (fire fighters,

Medical, Police…).In order to explain these steps, I use the architecture presented in figure 8;it has been proposed and implemented

in (Mintz-Habib, Rawat, Schulzrinne and Wu, 2005), as a VoIP based on SIP emergency

services architecture and prototype.

Identifying Emergency call

According to H. Schulzrinne, “Emergency services URI for the session initiation

protocol,” call/video packets to 9-1-1 can be identified by their destination and the

location of the call provided in the SIP URIs from the caller.Note that also progress works

are now being carried in order to standardize the national emergency number to be “sos”; this one will be the username part of the SIP

URI to represent an emergency call according to (Mintz-Habib, Rawat, Schulzrinne and Wu,

2005). Telephone URIs9 for conventional emergency numbers, i.e. “tel:911”, can be

aliased to the emergency URI “sos ”,either by a SIP UA or a SIP proxy,based on the location of the calling party.

Figure8. Emergency services system architecture (Mintz-Habib,et al.,2005)

Determine the Caller’s Location

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Identif yemergen

cy calls

Determi ne

location

Route call to

the correct

PSAP

Presen t call

to thecall-

taker

sip:[email protected]


The location information of the subscriber is essential for emergency call routing and

handling. The proxy server requires calling part’s location details to route calls to the proper PSAP, which in turn is supposed to provide emergency services to the caller.The

location information is required also in dispatching help to 9-1-1 callers. Since VoIP subscribers are nomadic,their location most

of the time are not easy to determinable.Some intelligent algorithm should be established.

Let’s consider a VoIP subscriber located in Edmond registered to IMS SIP server located

in Kigali,Rwanda,if the subscriber requests emergency services,the call should be routed

to the PSAP located in Edmond not in Kigali. Using the outbound proxy it is possible to

determine the caller’s MAC address. To determine this location the outbound can send

a DHCPINFOM query with the MAC retrieved from the packets it has received.

According to [8], it is possible for UA to determine its location by contacting the

Location Server.Note also that the MS with the use of SIP event notifications architecture

could subscribe to its own location using the location servers’ content.

Emergency Call Routing This step is implemented using Selective

Router in the current E9-1-1 network architecture,we have highlighted that using

SR in the 9-1-1 network create single point of failure; we need a way to provide

backups/redundant of these equipment. In the architecture presented in figure 8 DNS Naming

Authority Pointer Resource Records (NAPTR) are used to determine the correct PSAPs.

Call Presentation to the dispatcher in PSAPs The National Emergency Number Association

NENA has specified a list of features for presenting emergency calls to the call-takers

in PSAPs.Displays in PSAPs have to be able to display caller’s location on a map,log calls details into databases,store multimedia call

content, view call logs and generate statistics, and monitor currently active

calls and automatically distribute incoming calls to the available dispatchers.

Prototype 911system using VoIP

Voice over Internet Protocol (VoIP)has become a key technology of communication systems.

This section aims to the development of a VoIP system based on Asterisk that will work as UA agents and PROXY servers of an enhanced

emergency communication system . Previously we discussed necessities to deploy VoIP

systems such as IP networks, Signaling protocols i.e. SIP and how the current 9-1-1

system works to offer its daily functionalities services (call

identification, call routing, determine the location of the caller,and call presentation

to the dispatcher in PSAPs plus the drawbacks due to the fact that the current systems that

is still based on analogy transmissions).

I am proposing a prototype of emergency system based on ASTERISK;which “is a free and

open source framework for building communications applications and is sponsored

by Digium. Asterisk turns an ordinary computer into a communications server.

Asterisk powers IP PBX systems,VoIP gateways, and conference servers and is used by small

businesses, large businesses, call centers, carriers and governments worldwide) can be

used to test some of these requiredfeatures”10 and SIP softphone “this a software

(written programming java, C#...)for making phone calls over IP network using general

purpose computers,instead of using specific hardware.Most of the time softphone acts like

hardphones (CISCO IP phones or ALCATEL phones…) when runs on personal computer they

can be used to make telephone calls. Some appears like normal phones with smart

interactive interfaces (buttons, they can ring,they can display image or video calls...)

figure 10 show the look of a softphone. Note that those softphone in SIP architecture will be working as UA devices that make calls

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or requests to the 9-1-1 dispatchers through the SIP proxy server in this paper considered

as ASTERISK server.

In this prototype we are assuming that locations of people who are summoning for emergency help can be looked up using Dynamic

Host Configuration protocol (DHCP)which is a host specific services offered by many

routing devices or there network entities retrieved from Global positioning systems

that can query information from the SIP entities called LOCATION SERVER.

ASTERISK softPBX will be running on Ubuntu 12.04installed Oracle VM VirtualBox;which is free software that can be used to run many

operating system (OS) on one main OS. The purpose of Asterisk is for call routing it

will be situated in the CORE LAYER as the SIP proxy server in the. It will be handling

registrations process, location determination and providing advanced

telephony features like Voice and Video calls,Voicemail,Ring group,IVR (Interactive Voice Response), queuing, call parking &

transfer,and conferencing calls.

Figure 10: Softphone: Desire making a video call.

The reason of suggesting all of these

features are explained in the following paragraphs:

Deaf persons can use sign language to summon emergency first responders using SIP

entities that support video calls. In this setup the calltakers in the PSAPs will be able

to help the caller. Video calls can also be used in critical situations like when tele-

medicine assistance needs to be provided.The feature also can be used while report thieves

in stores or trying to present to the caller taker in PSAPs the current situation on the field such that he/she can plan necessities

for support (ASSESSMENT Of the Emergency CASE).Dispatchers in PSAPs will be using the video call feature to provide instruction on

how to handle emergency situations can behandled.

Conferencing calls also have to be introduced in the current emergency communication

system such that it will be possible to bridges many communication agencies; this is

due to the work coordination,information and resources sharing.

Call transfer or DID features must be offered in 9-1-1 systems such that anyone involved in

supporting (dispatchers,location specialist, defense agent)be reached at reasonable cost.

ASTERISK uses Perl scripts for routing

decisions. Asterisk will be configured to provide Interactive Response (IVR) services,

call queuing and playing announcements to people waiting in queue. To identify the proper destination of the call (appropriate

PSAP) Dynamic Name Server (DNS) will be used for lookups of the SIP URl in the invites sent

by the User Agent of the caller summoning help.

As we started above in the core of the emergency system applications like calls

distribution need to be provided such that the system will support multiple calltakers.

This prototype assumes that emergency call can be identified by looking up in the SIP

calls packets that has its destination to 9-1-1 or “SoS”.The presented softphone in FIGURE 10

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can be enhanced also in order to speed-up the emergency call dialing steps by adding

emergency call button such that anyone using the application can press the button that will ring 9-1-1 instead of dialing 3 times.

To determine the collect location of SIP UA, automatic look-up step in the Asterisk proxy

server has to be implemented. This process will be carried using DHCPINFM query of the

calling party MAC.This MAC address has to be carried in the SIP header of the packets calling an emergency call. Finally to complete the look-up; the SIP proxy will be

responsible for also find the current location of calls from a Public Switched

Telephone Network to VoIP network by retrieving the source phone number. Using

this number SIP proxy can contact the Home Location Register (HLR) or the Visitor Location Register) an entities in

Telecommunication networks that store the current locations of the subscribers. I am

assuming that they should be tracing mechanism that can be adopted to trace retrieved MAC address in the emergency call

packets such that the location of the caller UA be determined. After finding the location of caller via SIP INVITE requests the proxy

server will encapsulate the information (locations details).As described in FIGURE 11

a SIP UA will register to the network and then send a call to 9-1-1. This call will be first

directed to the outbound proxy server will used DHCP INFOM to retrieve the location of

the caller and encapsulate the call for further treatment in the core and routing decision issues.

Figure 11:identify emergency calls,determine the caller location and forward the call to PSAP Server.

Emergency calls will be routed to the appropriate PSAPs based on the location of

the caller. DNS lookups are preferred here that will be using MAPInfo’s ENVINSA which is

a “location platform providing a broad suite of capabilities that can be applied to solve

anywhere from a single location need to a wide variety of problems11 ” the Envinsa will be

implemented in the HLR register (see figure 11) It will help to find appropriate PSAP

information based on geographic location. Scripts running on the SIP proxy server have

to able to queries the HLR (based on Envinsa) in order to route call to the correct PSAP.

Locations information along with the calls must be routed to correct PSAPS in the

architecture presented in figure 11. SIP and signaling protocol can be used to

deliver emergency services over IP networks. This means that based on gateways call from

other protocols. The presented architecture can be enhanced by adding gatekeepers and

gateways such that other protocol may be supported.

Implementing emergency Notification s

The Emergency notification subsystem in 9-1-1 network architecture is responsible of

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allowing government officials notifying the public about an emergency and the required

measures to take as preventions mechanisms. Those notifications maybe exchange between

from national law enforcement agencies to police stations, public first responders,

government agencies, private first responders agencies or form the government to

public. In this section I am proposing an emergency alerting system based SIP that can

be used instead of the existing one based on PSTN that is not scalable, slow and lacking

advanced features. The current Emergency Alert System (EAS) has

been revised by the Federal Communications Commission from Emergency Broadcast System

that was only configured to be used by the President, to an the EAS that is able to be

called up by local &states authority. The EAS system use Frequency Modulation (FM) and Amplitude Modulation (AM)and TV stations

to broadcast information. Every station is supposed to listens to 2 other stations for

emergency ALERTS and automatically broadcasts them for its geographical area.

The component of this alert is the alert tone, audio-messages and an end of message

indication.These old systems also use sirens for notifications but offer limited

information.

The advanced architecture of EAS can be build based on SIP event notification system. I am

assuming that subscribers to 3G/4G networks have UA agents on their devices that can

process SIP event notifications. The architecture of this system has its basis

on where the information come from; first of all alerts are originated from the government

agencies to states and local governments and the public; all information will be shared

between these layers. This is an hierarchical i.e.figure 12 model as

the users registered to top layer serves as servers to users registered to them (low

lever).

Figure 12:Alert hierarchical model

Figure 13:Alert exchange info As we see in Figure 13 the UA sends subscribe

request in order to get a events notifications.In this schema the subscriber

is allowed to send the media of the information is capable of processing either

audio or video. As we see this notification will provide

greater reachability, speed and data integrity.

Conclusion and Recommendations

In this paper we revised mechanism that can be used to provide emergency services using IP

Networks;we have also discussed how VoIP can be used to provide Emergency services and

then finally we discussed how Internet can be used to alert the public about the pending or

current emergency situations. We recommend that future studies be carried in order to study all requirements needed such that this

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US Fed Govt.

Oklahoma

Edmond

Citizens


prototype be deployed on the national level.

References [1] J. Rosenberg, H. Schulzrinne, G. Camarillo,

A.R.Johnston,J.Peterson,R.Sparks,M.Handley, and E. Schooler, “SIP: Session initiation

protocol,”RFC 3261,June 2002. [2] Dale Hatfield, Brad Bernthal, and Phil

Weiser, Health of the US 9-1-1 System, sponsored by the 9-1-1 Industry Alliance,2008,

Part II, “Section B. Background and Evolution of Our Nation’s 9-1-1 System,” page

20Report athttp://www.911alliance.org/9IA_Health_of_US_911%20_2_.pdf .

[3] NENA, Next Generation Partner Program, A Policy Maker Blueprint for Transitioning to

the Next Generation 9-1-1 System: Issues and Recommendations for State and Federal Policy

Makers to EnableNG911,September2008,page2.Reportathttp://www.nena.org/sites/default/files/NG911PolicyMakerBlueprintTransitionGuide-Final_0.pdf [5] Peter Dely,“Adaptive Aggregation of Voice

over IP in Wireless Mesh Networks”, Karlstad University.Approved, 2007-06-07

[6] Butt Muhammad Faisal Nazir” Performance Analysis of Asterisk VoIP Server with

Static and Real-Time Database Call Processing” page 28,2006

[7] Jan Janak “SIP Introduction”, FhG FOKUS, 2003 [8] H. Schulzrinne, “Emergency

services URI for the session initiation protocol,”draft-ietf-sipping-sos-01,Internet

Draft,Feb.2004,work in progress. [9] H.Schulzrinne,“The tel URI for telephone

numbers,”RFC 3966,Dec.2004 [10] http://www.asterisk.org/get-started

[11]http://www.empower.com/pages/products_envinsa.htm

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http://www.empower.com/pages/products_envinsa.htm

http://www.empower.com/pages/products_envinsa.htm

http://www.asterisk.org/get-started

http://www.911alliance.org/9IA_Health_of_US_911%20_2_.pdf

http://www.911alliance.org/9IA_Health_of_US_911%20_2_.pdf

Study Mechanism for Providing Emergency Services using VoIP/IP Network (Match 2014

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