Product Requirement Specification

7/31/2019 Product Requirement Specification

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Product Requirement Specification for Mobile Client V2.1

Company Confidential Page 1 of 22


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Product Requirement Specification for Mobile SIP Dialer V1.0

01/09/09 1.0 Initial Draft Sin Ming Esther, Irene, Steven Watt, Yusnidah,Timothy, Kelvin, Beng Teik

25/11/09 2.0 Update the following

1. Section 4.1, 4.3, 4.6, 4.8, 4.12

Add the following

1. Section 4.13 to 4.24

Sin Ming Wee Sin, Soon Liong

30/11/09 2.1 Update the following

1. Section 4.9m 4.16, 4.17

Add the following

1. 4.2.22, 4.20-4.26

Sin Ming



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1. Introduction......................................................................................................................5

1.1 Purpose...................................................................................................................51.2 Intended Audience...................................................................................................5

1.3 Specification Scope..................................................................................................51.4 Reference................................................................................................................5

2. Product Description...........................................................................................................6

2.1 Product Overview.....................................................................................................62.1.1 Usage Mode...............................................................................................................62.1.2 Business Model..........................................................................................................6

2.2 System Design Requirement.....................................................................................62.3 Operating System and Environment.........................................................................62.4 Assumptions and Dependencies...............................................................................6

3. Interface Requirements ....................................................................................................7

3.1 User Interfaces........................................................................................................73.2 Hardware Interfaces................................................................................................73.3 Software Interfaces..................................................................................................73.4 Communication Interfaces........................................................................................7

4. Features & Functional Requirements..................................................................................8

4.1 Support Device (Priority 1).......................................................................................84.2 Mobile Client Features (Priority 1).............................................................................8

4.2.1 Making and receiving VoIP calls (Priority 1)................................................................84.2.2 VoIP over 3G and WiFi (Priority 1)..............................................................................94.2.3 VoIP over GPRS (Priority 1)........................................................................................94.2.4 RFC 2833 and SIP Info DTMF Support (Priority 1).......................................................94.2.5 Phone Book (Priority 1)..............................................................................................94.2.6 View Call Log OK (Priority 1).....................................................................................94.2.7 Loudspeaker OK (Priority 1).....................................................................................104.2.8 Configurable Seamless trigger of Company VoIP call. (Priority 1)..............................104.2.9 Remember and Auto Sign in to Access Point (Priority 1)............................................114.2.10 Client upgrade notification (Priority 1)....................................................................114.2.11 SIP Proxy Failover (Priority 1).................................................................................114.2.12 Display proprietary error message(Priority 1) .........................................................124.2.13 Account Balance Display (Priority 1).......................................................................124.2.14 Call Duration Allowed (TalkTime) Display (Priority 1).............................................124.2.15 Live timer during call (Priority 1)...........................................................................124.2.16 Last 5 number Redial (Priority 1)............................................................................12

4.2.17 Low balance alert (Priority 1)..................................................................................134.2.18 Call Hold (Priority 1)...............................................................................................134.2.19 Codec Support (Priority 1)......................................................................................134.2.20 VoIP Call History (Priority 2)...................................................................................144.2.21 Auto Reply (removed)............................................................................................144.2.22 Caller ID Display (Priority 1)...................................................................................144.2.23 Auto Start (Priority 1).............................................................................................14

4.3 Serving Advertisement when user make call OK (Priority 1)....................................154.4 Fund Transfer (Must have) OK (Priority 1)..............................................................154.5 CallBack (Priority 1)................................................................................................164.6 SMS (Priority 1)......................................................................................................164.7 Anti-blocking mechanism (Priority 1)......................................................................17

4.8 Presence/IM (Priority 2).........................................................................................17



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4.9 Voice Mail (Priority 1).............................................................................................174.10 Call Forward (Priority 1).......................................................................................184.11 Call Transfer (Priority 1).......................................................................................184.12 Conferences (Priority 1)........................................................................................184.13 Centralized Phone book(Priority 2) .......................................................................194.14 Multiple Language Support (Priority 2)..................................................................19

4.15 Value Add Service Content (Priority 1)...............................................................194.16 Interop with Other Client (Priority 2)....................................................................204.17 Fault Reporting from Dialier (Priority 1)................................................................204.18 Contact Us Email (Priority 1).................................................................................21

5. Reporting and Monitoring Requirements..........................................................................21

6. Other Nonfunctional Requirements..................................................................................21

6.1 Performance Requirements....................................................................................216.2 Safety Requirements..............................................................................................216.3 Security Requirements...........................................................................................216.4 Quality Requirements.............................................................................................216.5 Business Rules.......................................................................................................216.6 User Documentation...............................................................................................216.7 Software statutory & regulatory requirement..........................................................216.8 Is a Trial version required? ....................................................................................216.9 3rd Party product Compatibility List........................................................................22



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This Product Requirements Specification (PRS) defines the

requirements for Mobile Client. This PRS states the functions andcapabilities of the Product and the constraints that it operates under. Itwill be used to communicate the specification clearly among variousstakeholders within Company. This document will also be the basis forall subsequent project planning, design, and coding, as well as thefoundation for system testing and user documentation.

The intended audiences of this PRS are people involved in thefollowing activities: Sales, Deployment, Support, Billing, and

Development & Testing

This document contains:

This PRS is to define the essential features for the development ofCompany Mobile VoIP service.

Other documentation related to this product (but not part of thisproduct) is:



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2.1.1 Usage Mode 1. The user goes to a self registration web page to signup for this mobile

VoIP service. OK

2. Upon successful registration, the system will trigger a sms to thecustomers handphone. OK Customer must also select mobile phonemanufacturer and model so that mobile platform can be determinedand appropriate link could but sent to customer.(sin ming OK)

3.

4. The sms shall contains a link, in which when click, will download theclient to the user phone and install the client into the users phone. OK

5. When the user first starts the client, he will be prompt to enter hisusername and password. OK

6. Prior to making the VoIP call, the user must first logon to his WLAN.OK

7. To make the VoIP call, the user dial the destination PSTN phonenumber and then initiate a internet call. OK

8. The sip proxy shall authenticate the user and check if he has sufficientbalance to make the internet call. OK

9. If so, the sip proxy shall connect the call.Ok

10. If not, the sip proxy shall terminate the call.OK

11. The sip proxy shall rate the call after the user hangs up. OK

2.1.2 Business Model

All user shall have a pre paid account.

All on-net call shall be free.

All off-net call shall be charged based on the rate table.

None

The SIP Proxy shall operate on Linux

None



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TBD

None

The users account is authenticated by Company RTBS. Therefore theSIP Proxy must be able to understand the RADIUS Protocol.

None



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The followings are the platform to be supported

1. Nokia Phone

a. Symbian 2nd Edition feature pack 1, 2nd Edition feature pack 1and 2nd Edition feature pack 2 We do not have 2nd edition andwe never intended to have it as our roadmap because wethink 2nd edition phones are rarely used and have otherissues also. (sin ming ok)

b. Symbian 3rd Edition, 3rd Edition feature pack 1 and 3rd Editionfeature pack 2 OK

c. Symbian 5th Edition OK

2. Window Mobile OK

a. Window Mobile 5

b. Window Mobile 6

c. Window Mobile 6.1

d. Window Mobile 6.5

3. iPhone OK

a. 2G

b. 3G

c. 3Gs

4. Android OK

a. Version 1.5 onwards

5. Blackberry *

6. LG *

7. Sony Ericsson*

8. Samsung*

For those device (*) which cannot support VoIP, callback shallreplace the voip call.

The client shall support RFC 3621.

4.2.1 Making and receiving VoIP calls (Priority 1)

The client shall be able to make to1. PSTN number (Off net call) OK



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2. Another client (On net call) OK3. MRTalk (Company Soft dialer) (On net call using alphanumeric user

id) OK

The client shall be able to receive call from

1. Another client (on net call)OK2. MRTalk (Company Soft dialer)OK

4.2.2 VoIP over 3G and WiFi (Priority 1)

The client shall be able to connect, make call and receive call via 3Gand WiFi by selecting the appropriate access point.OK

4.2.3 VoIP over GPRS (Priority 1)

The Client shall be able to connect, make call and receive call via the14kbps low bandwidth of GPRS.

The client shall detect that the user is using the GPRS network andadopt a different strategy to accommodate the low up ramp bandwidth ofthe GPRS network.

The client shall pack 4audio frame in a single audio packet send overCompanys own proprietary 2 byte header (in place of RTP header).

The client shall accept the normal RTP packets for the incoming audio

since there is sufficient bandwidth for the down ramp for GPRS. OK.Please share the RTP packet header info. And why not adopt the sametechnique even in WIFI network or 3G?(sin ming. The strategy for compacting multiple frame in 1 audio packetcan degrade audio quality. Therefore I would like to use it only for GPRSand not compromise the quality for WiFi and 3G)

4.2.4 RFC 2833 and SIP Info DTMF Support (Priority 1)

Mobile Client shall support RFC 2833 and SIP Info DTMF. OK

4.2.5 Phone Book (Priority 1)Mobile Client shall be able to access the Nokia Phones native phone.From the Client phone book, the user shall be able to search and makeVoIP call.OK

4.2.6 View Call Log OK (Priority 1)

All the call made using the Company Client shall be recorded.The user shall be able to choose to view the call history through theclient.

The user shall be able to set the number of call detail he wants to view.



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(Company shall provide a http api to get the call history for that account).

4.2.7 Loudspeaker OK (Priority 1)

When the call is connected, the user can switch to loudspeaker modeand back to handset mode.

4.2.8 Configurable Seamless trigger of Company VoIP call. (Priority1)

Mobile Client shall run in the background. It will automatically comes tothe foreground when the user makes a VoIP Call..

The user can set the default call type of the client

Internet Call When the user clicks on green make call button, if client is registeredwith sip proxy, the client will automatically makes a voip call. If notregistered, it will make a normal gsm call.

Always ask When the user clicks on the green make call button,the Company client will prompt the user if he wants to make a GSM orVoIP Call.If he selects the GSM call, the normal GSM call shall be made.

If the user selects the VoIP Call,If the mobile client is registered with the sip proxy, a Company VoIPCall shall be made.If the client is not registered to Company VoIP, the client shallautomatically make a GSM call without asking. If we are decidingbetween VOIP or GSM call on the basis of the fact if dialer is registeredor not what is the point of asking from user? Please explain where thisdialog will fit.(sin ming. The dialer shall only prompt the user if it is registered with oursip proxy. The user might choose to make a GSM call if he is calling alocal call.

If the dialer is not register, there is no point asking since it cannot makea VoIP call. In this case, it will fall automatically to the GSM call).

GSM Call When the user clicks on the green make call button, the client shallmake a normal GSM call.

The client shall be able to intercept call make from1. The native phone call home menu2. The native phone phonebook

3. The native phone call record



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a. Dailed numberb. Received numberc. Missed Call

4.2.9 Remember and Auto Sign in to Access Point (Priority 1)

1. The mobile client shall store the access points which the user has

successfully login before. OK

2. If the user comes into this wifi zone or 3G network, the mobil client

shall automatically logon to the access point and registered with the sip

proxy. OK but wifi password will be prompted. (sin ming This should

only happens if that access point is not already added into the phone)

3. WiFi access point shall have priority over the 3G access point.

4. The user can manage these access point list. He can OK after saving

changes will be applied after application reboot. (sin ming Which

platform has this behavior? Our symbian dialer allows to add and

remove the access point without needing to reboot the phone. We

should not need to do so.)

a. Remove the access point from the list.

b. Add access point to the list.

c. Change priority of the access point in the list.

4.2.10 Client upgrade notification (Priority 1)

The user shall be inform if there is an upgrade of the client.

There shall be 2 type of upgrade. OK1. Mandatory upgrade in which the user cannot use the service until he

upgrade the client.

2. Recommended upgrade. In this case, the user shall be inform of thatthere is a new client. The user can still use the service even if he didnot upgrade the client.

(Company shall provide upgrade mechanism)

4.2.11 SIP Proxy Failover (Priority 1)

The client shall have 4 sip proxyies configured. It failover to the next sip

proxy if the current sip proxy cannot be contacted or return an error I



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suggest we use only one proxy and this proxy we get from one singlehttp response which was used to tell us if upgrade is available atstartup. From backend you change proxy whenever its down. Suggestme (sin ming, The sip proxy failover is necessary to avoid service downtime in case our sip proxy is down. Upgrading of the dialer to change

the IP might be a bit too slow.)

4.2.12 Display proprietary error message(Priority 1)

The client shall be able to display proprietary error message as returnby the sip proxy.The sip proxy shall return the error message in the Reason-Header. ok

4.2.13 Account Balance Display (Priority 1)

The client shall display the users account balance on the UI of theclient.

The sip proxy shall return the account balance in a proprietary header inthe 200 OK message (registration response). OK so there will not anyhttp URL called to fetch balance? (Sin Ming. The account balance iscurrently return by our sip proxy in the registration.)

4.2.14 Call Duration Allowed (TalkTime) Display (Priority 1)

When the user clicks to make call, the client shall on its UI the allow call

duration.

The sip proxy shall return the talktime for the call in a proprietary headerin the 180,183 and 200 sip mesagemessage OK

4.2.15 Live timer during call (Priority 1)

After the call is connected, the client shall show a live which incrementevery seconds. The live timer shall be displayed in the following formathh:mm:ss OK

4.2.16 Last 5 number Redial (Priority 1)

The client shall keep the last 5 called number. The user can clicks onthese number to redial them.last 5 dialed numbers? Do you mean callhistory (which comes via your API?) or nokia phones dialed number list.According to functionality we will capture calls from contact, local nokiacall log etc anyway. Please explain. (sin ming Can you put the voip callmade from the dialer into the nokia phone dialed list? Basically oursymbian dialer allows calls to be made from the nokia native phone (in

the call intercept mode) and also from the dialer itself. We can probably



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combined this feature with the 4.2.20 VoIP Call History. Basically allowsthe user to click on the numbers in the Call History to make call)

4.2.17 Low balance alert (Priority 1)

The client shall display a warning message if the users account is lessthan a specified amount.

The SIP Proxy shall send the alert in the 200 OK message in responseto the client registration message.OK full format needed. (sin ming. Wewill provide the details on the proprietary header later)

4.2.18 Call Hold (Priority 1)

The client shall be able to put on hold when it receives a gsm or voipcall.

The client should be able to switch between the active call and callwhich is on hold.

If the user is engage in a gsm call and a call voip call came in, in whichthe user answer the call, the gsm call shall be put on hold.

Ok but if customer is using GPRS and an incoming GSM call comes symbiandisconnects GPRS anyway. We will try to see what control can have to identify suchevent. For 3G and WIFI hold can be implemented but we will decide the exactfunctionality after doing some test work. The reason is when a VOIP call is in progress

on WIFI and an incoming call comes we will have to see how events are captured andallowed by symbian.(sin ming. It can be done. We can put the VoIP call on hold)

4.2.19 Codec Support (Priority 1)

The codec of preference is in the following order1. G7292. G723 not available (sin ming understood)3. iLBC4. GSM

5. AMR

The client shall as far as possible use codec that is supported by thedevice and avoid using soft codec. OK presently we are using devicespecific codecs only. In some devices iLBC and g729 are there (Likesymbian 3rd edition and 5th edition) but in most of the devices g729 is notthere. AMR is the only codec that is present in iPhone and Android too.In Blackberry also AMR is there.We have choice of implementing ITU g729 or free iLBC codec or wecan opt for third party hardware optimized codecs for the platformswhere such codecs are avialble. GIPS, Voiceage and SPIRIT DSP are

options.



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4.2.20 VoIP Call History (Priority 2)

The client shall keep a record of all the voip calls made, received andmissed by the client.These call records shall be view in the client and also be inserted intothe native phone call history (if possible).OK I think for all kinds of call we will get output from http APIs? (sinming. Actually I am thinking of getting the phone to keep a record of thedialed number, received call and missed call. We use the http api for the4.2.6 to get the duration and charge for the call. Again, we probably cancombine this feature with 4.2.6 and 4.2.16)(Note that the sip proxy will indicates to the client if this is a missed call.

This is because our sip proxy do call forking to multiple endpoint. So ifthe user picks up the call in another device, that call should not bemarked as a missed call. I will provide the detail later).

4.2.21 Auto Reply (removed)

The user be able to set the client to be an auto reply mode and also themessage which is to be send to the caller. Please explain this feature. Ithink you already have a voice mail box. It would be good if Do notdisturb flag can be set in your server and from there itself it can take tovoicemail. In many platforms implementing voice message (Local )would be an issue, if it is server based it will be for all platforms . Pleasecomment (sin ming after consideration, we decide to remove thisfeature)

4.2.22 Caller ID Display (Priority 1)

The client shall be able to display the caller ID when the client receivesan on net call. OK. To be more clear. In every incoming call Invite wewill have caller id like Manoj whereManoj is caller id. Am I correct? (sin ming Yes)

4.2.23 Auto Start (Priority 1)

When the user first starts the client, the client shall prompt him for theuser user ID (MRTalk ID) and password.OK

The client shall also ask the user if he wants to auto start the clientwhenever the phone starts up.OK

If the user selects yes, the client shall be started whenever the phone re-starts.OK

The user can change this selection through the client settings.OK



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The client shall display advertisement to the user before and after thecall. The advertisement display shall be based on the user profilecaptured when the user signup for the MVoIP service.When the user clicks to make call,1. The client shall query Company advertisement server (using a http

api provided by Company) .

2. The advertisement server shall return 2 banner URL and an audioadvertisement to play.

3. The client open a browser within the client and display the banner 1URL.

4. The client shall then send an sip invite message to the sip proxy,passing to it the audio advertisement to pay in the proprietaryheader.

5. The sip proxy shall play the audio advertisement and then connectsthe call.

6. When the call is ended, the client shall display another banner 2within the embedded browser.

7. Note that the client shall provide click through for both the banner. Inanother words, the client shall open the browser and direct thebrowser to the URL assosicated with the banner.

(Details on the ads server interface shall be provided by Company)

(Note that there is a possibility that for some premium customer,

Company might choose not to serve advertisement to. In this case, thead server will not return any banner. Under this circumstances, theclient need not embedded the browser in the dialer).

The client shall have the user interface for performing fund transfer.The user can1. Transfer to Another Company Account

For this feature, the user can transfer balance from his account in theclient to another account specified. The user can specify the amounthe wants to transfer.



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2. Fund transfer to 3rd party.

3. RechargeThe user can transfer money from other account to his account. The

user can specify the amount he wants to transfer. Default being all.(Company shall provide http api to allow the client to do the fundtransfer function.)

The client shall be able to set the phone to make a voip call or a callbackcall.

2 mode of callback shall be supported which can be set by the user.Please explain where can we set this option? In settings? Do you mean

something like choose between Voip call and Callback (sin ming.Yes, the user can set the default call type. Whether he wants to alwaysmake a VoIP call or a callback call).1. Callback trigger via IP. The callback is triggered by sending a http

request to Company callback proxy.

2. Callback trigger by dialing to an access number.a. When the user clicks to make call, the client shall make a call

to an access number.b. Company callback proxy shall first check that the ani of the

incoming call has been registered with Company.c. If so, Company client shall terminates the incoming call and

then connects to the callers ani. I think you mean callbackserver will disconnect incoming call (Not Company client) (sinming yes. Sorry about the mistake)

d. The client shall auto pick up the call and automatically sendthe dtmf of the destination he wants to call.

e. Company callback proxy shall then connects the client to thedestination.

(Detail design for this service shall be provided by Company) Please

provide detailed design (sin ming. We are currently working on it. Willprovide the detail in the earliest possible moment)

The client shall be able to send sms.

The client shall have a editor for the user to enters his select thedestination he wants to send the sms to.The user can also set the client to operate in a sms intercept mode.

The sms message is send via Company sms server using the sip



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message.(Details on the sms server interface shall be provided by Company.) OK

Currently there is ISP which block VoIP calls.The client shall implement Companys anti-blocking mechanism whichinvolves the encryption (using xor) of the signaling and audio.(Details of this anti-blocking shall to be provided by Company) OK

The client shall have all the Instant Messages features (IM) such as1. Presence2. Click on user to

a. Send sms

b. Chatc. Make call3. File Transfer (for transferring of photos etc).4. The IM client client shall be able to interop with other IM such as

MSN, Yahoo, GTalk etc.The IM shall use XMPP protocol

As discussed on phone I am giving you an overview of how IM should beimplemented.Implement Simple supported server or implement Simple in your SIPproxy itself (instead of directly accepting XMMP messages from client).

This way Company client to client Voice also supported using a SIPmessage. There are jabber server plugins, which are based on XMMP.These plugins will give you IM interfaces with MSN, Gtalk and yahoo etc.This way Company client will add thirdparty user

Another way is to implement only jabber server and client will talkindependently to jabber server in case user selected IM however in caseof voice call SIP proxy will be used.

Let me know what you think about it. After I get more information aboutthe kind of network you are running I can assist more.

If you already have some information, For voice call between Companyclient and yahoo/gtalk or MSN, please share with us to speedup.

The client shall be notify of any voicemail for him (using messagewaiting indicator MWI). This message list will be fetched only atapplication startup or after certain intervals?

The user can click to access his voicemail.



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(Details shall be provided by Company on the voicemail interface) OKmore questions may be there after we receive details.

(sin ming. Our current system uses subscribe and notify to get the voicemailAlert.The flow is briefly describe as follow)

The sip proxy shall inform the client the voicemail server IP for this client in theregistration 200 OK message. The client will then send a sip subscribe message tothe sip proxy with the voicemail server in the Route field of the subscribe message.When there is an voice mail, the sip proxy shall send a SIP Notify message of thevoicemail. The client shall display on the UI that a voicemail is present. When theuser clicks to listen to the voicemail, the client shall make a VoIP call to a certainnumber (to the voice server) with the voicemail server IP in the Route field.)

The client shall provide an easy mechanism for the user to set/change

his call forward number.(Details shall be provided by Company on the api for setting of callforward number) I think some screen could be provided in Settings toset forward number. I guess you will provide some http API to set it toyour server? (sin ming. Yes, we will)

The client shall be able to perform call transfer between 2 VoIP call.

The client shall be able to put the first call on hold (this first call can be

a call made by him or received by the client).

Then make a second call out.

And transfer call 1 to call 2.

(Details shall be provided by Company on the signaling for calltransfer)

I think this would be a difficult thing to achieve as in most of the platforms we plan use devicespecific APIs for all audio handling for better performance. To switch between calls would requirecustom audio mixer.But if we decide to use third party media engine like SPIRIT DSP we can do that easily I think. Pleasecomment.

The user shall be click to participate in a multiply parties conference bycalling to Company conference server.The conference server shall return in a proprietary header1. Status of the conference2. A list of the participants (MRTalk ID for Company client or caller ID if

the participant call from PSTN).3. The time in which each participant has been on the conference.

The client shall display the above information during the conference



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call.(Detail shall be provided by Company on interface with their conferenceserver) ok

The client shall be able to load all his contacts from his native phone tothe centralized phonebook.The client shall also perform a full sync with the centralized phonebook.The client shall be able to perform conflict dispute.SyncML protocol is used for the uploading and synchronizing of thecontacts.(Details on the centralized phone book interface shall be provided byCompany).OK we will wait for the details.

The client shall support multiple language. Some of the language to besupported are1. English2. Mandarin3. Hindi4. Malayalam5. Tamil6. ArabicThe user select the language on the client .

When set, all the text in the client shall be display in the languageselected.

The IVR shall also be display in the language selected.

The client shall send the user selected language to the sip proxy whichwill instruct the IVR to play the language selected.

(Details on the provided later).OK

Company is currently working with spice to provide content such as1. Music2. Chanting3. Sport etc

The user shall be able to access the content easily from the mobileclient.

Spice Content Server shall provide the list of contents (this list canchange dynamically).



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The client shall be able to interface with spice content provider server toget the list (TBD) and display to the user.

The user shall be able to select and hear the content.

The user can also select to download the song into the client (in mp3format) and play it using the clients native media player.

Details shall be provided by Company. More details required for this tocomment.

The client shall be able to interwork with other major IM client both interms of voice and IM functions.IM such as

1. Yahoo2. Skype3. MSN4. Gtalk

The interop is done at the Company backend.

The client must provide an interface for the user to specify the IM typewhich the friend his using.

Details to be provided by Company.

Ok I already explained on what we think about IM server infrastructure.After network is decided (Please ignore if you have already decided)

The user can report fault from the client.

There shall be 2 mode of fault report.1. The user can trigger a fault report from the client option2. The sip proxy returns a proprietary message header in its response

message. The client shall display the error and ask the user if he

wants to report this fault. The user can choose yes or no.

In either cases, the client shall collect the call information and send it toCompany fault report server via a http api.

The type of information to be collected and the httpd api shall be defineby Company.

When we say fault do we mean Call failure reasons? if yes do you mean that on every call failureNot found, Temporary failure we have to ask user to submit information? (sin ming. We have notexactly define the nature of the report yet. But at a minimum, I need to know the username, thedestination he calls, the callID of the call, all the transaction between the client and all our servers

and also some audio report such as percentage of loss packet, jitter information etc)



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The clients shall have Company Contact Us email in which the user canaccess to easily.

Basically the user just click the Contact Us options and Companyssupport center email is presented to him.

If you can provide me an http URL which can save some content directly in your ticketing system, itwould be very easy. This way we will just provide an option for contact us screen (like typing SMS)and submit. Thats it. Please suggest (sin ming. We can hardcode the email first on the dialer).

Call Setup Within 10 seconds

Able to handle 2000 registration

Able to handle 500 concurrent calls

None

Customers information must be kept secure.

Call Rated correctly for on-net and off-net call.

Transaction logs and error logs during call

Audit trial which logs down all the activities of the customer careengineer.

TBD

System Installation Guide

None

(Delete as applicable)

(If YES) N0



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What are the 3rd Part product used and their compatibility with theproduct?


Documents

Product Requirement Specification