Product Requirement Specification for Mobile Client V2.

Mobile Client
Product Requirements Specification
Version <2.1>

Company Confidential

Page 1 of 22

Product Requirement Specification for Mobile SIP Dialer V1.0

Revision History
Date 01/09/09 25/11/09 Version 1.0 2.0 Initial Draft 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 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 Description Author Sin Ming Sin Ming Reviewer Esther, Irene, Steven Watt, Yusnidah, Timothy, Kelvin, Beng Teik Wee Sin, Soon Liong

Company Confidential

Page 2 of 22

Product Requirement Specification for Mobile SIP Dialer V1.0

Table of Contents
1. Introduction......................................................................................................................5 1.1 1.2 1.3 1.4 Purpose...................................................................................................................5 Intended Audience...................................................................................................5 Specification Scope..................................................................................................5 Reference................................................................................................................5

2. Product Description...........................................................................................................6 2.1 Product Overview.....................................................................................................6 2.1.1 Usage Mode...............................................................................................................6 2.1.2 Business Model..........................................................................................................6 2.2 System Design Requirement.....................................................................................6 2.3 Operating System and Environment.........................................................................6 2.4 Assumptions and Dependencies...............................................................................6 3. Interface Requirements ....................................................................................................7 3.1 3.2 3.3 3.4 User Interfaces........................................................................................................7 Hardware Interfaces................................................................................................7 Software Interfaces..................................................................................................7 Communication Interfaces........................................................................................7

4. Features & Functional Requirements..................................................................................8 4.1 Support Device (Priority 1).......................................................................................8 4.2 Mobile Client Features (Priority 1).............................................................................8 4.2.1 Making and receiving VoIP calls (Priority 1)................................................................8 4.2.2 VoIP over 3G and WiFi (Priority 1)..............................................................................9 4.2.3 VoIP over GPRS (Priority 1)........................................................................................9 4.2.4 RFC 2833 and SIP Info DTMF Support (Priority 1).......................................................9 4.2.5 Phone Book (Priority 1)..............................................................................................9 4.2.6 View Call Log OK (Priority 1).....................................................................................9 4.2.7 Loudspeaker OK (Priority 1).....................................................................................10 4.2.8 Configurable Seamless trigger of Company VoIP call. (Priority 1)..............................10 4.2.9 Remember and Auto Sign in to Access Point (Priority 1)............................................11 4.2.10 Client upgrade notification (Priority 1)....................................................................11 4.2.11 SIP Proxy Failover (Priority 1).................................................................................11 4.2.12 Display proprietary error message(Priority 1) .........................................................12 4.2.13 Account Balance Display (Priority 1).......................................................................12 4.2.14 Call Duration Allowed (TalkTime) Display (Priority 1).............................................12 4.2.15 Live timer during call (Priority 1)...........................................................................12 4.2.16 Last 5 number Redial (Priority 1)............................................................................12 4.2.17 Low balance alert (Priority 1)..................................................................................13 4.2.18 Call Hold (Priority 1)...............................................................................................13 4.2.19 Codec Support (Priority 1)......................................................................................13 4.2.20 VoIP Call History (Priority 2)...................................................................................14 4.2.21 Auto Reply (removed)............................................................................................14 4.2.22 Caller ID Display (Priority 1)...................................................................................14 4.2.23 Auto Start (Priority 1).............................................................................................14 4.3 Serving Advertisement when user make call OK (Priority 1)....................................15 4.4 Fund Transfer (Must have) OK (Priority 1)..............................................................15 4.5 CallBack (Priority 1)................................................................................................16 4.6 SMS (Priority 1)......................................................................................................16 4.7 Anti-blocking mechanism (Priority 1)......................................................................17 4.8 Presence/IM (Priority 2).........................................................................................17

Company Confidential

Page 3 of 22

Product Requirement Specification for Mobile SIP Dialer V1.0 4.9 Voice Mail (Priority 1).............................................................................................17 4.10 Call Forward (Priority 1).......................................................................................18 4.11 Call Transfer (Priority 1).......................................................................................18 4.12 Conferences (Priority 1)........................................................................................18 4.13 Centralized Phone book(Priority 2) .......................................................................19 4.14 Multiple Language Support (Priority 2)..................................................................19 4.15 Value Add Service Content (Priority 1)...............................................................19 4.16 Interop with Other Client (Priority 2)....................................................................20 4.17 Fault Reporting from Dialier (Priority 1)................................................................20 4.18 Contact Us Email (Priority 1).................................................................................21 5. Reporting and Monitoring Requirements..........................................................................21 6. Other Nonfunctional Requirements..................................................................................21 6.1 6.2 6.3 6.4 6.5 6.6 6.7 6.8 6.9 Performance Requirements....................................................................................21 Safety Requirements..............................................................................................21 Security Requirements...........................................................................................21 Quality Requirements.............................................................................................21 Business Rules.......................................................................................................21 User Documentation...............................................................................................21 Software statutory & regulatory requirement..........................................................21 Is a Trial version required? ....................................................................................21 3rd Party product Compatibility List........................................................................22

Company Confidential

Page 4 of 22

Product Requirement Specification for Mobile SIP Dialer V1.0

1. Introduction
1.1 Purpose
This Product Requirements Specification (PRS) defines the requirements for Mobile Client. This PRS states the functions and capabilities of the Product and the constraints that it operates under. It will be used to communicate the specification clearly among various stakeholders within Company. This document will also be the basis for all subsequent project planning, design, and coding, as well as the foundation for system testing and user documentation.

1.2

Intended Audience
The intended audiences of this PRS are people involved in the following activities: Sales, Deployment, Support, Billing, and Development & Testing

1.3

Specification Scope
This document contains: This PRS is to define the essential features for the development of Company Mobile VoIP service.

1.4

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

Company Confidential

Page 5 of 22

Product Requirement Specification for Mobile SIP Dialer V1.0

2. Product Description
2.1 Product Overview
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 the customers handphone. OK Customer must also select mobile phone manufacturer and model so that mobile platform can be determined and appropriate link could but sent to customer.(sin ming OK) 3. 4. The sms shall contains a link, in which when click, will download the client 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 his username 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 phone number and then initiate a internet call. OK 8. The sip proxy shall authenticate the user and check if he has sufficient balance 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.

2.2 2.3

System Design Requirement

None

Operating System and Environment

The SIP Proxy shall operate on Linux

2.4

Assumptions and Dependencies

None
Company Confidential
Page 6 of 22

Product Requirement Specification for Mobile SIP Dialer V1.0

3. Interface Requirements
3.1 3.2 3.3 User Interfaces
TBD

Hardware Interfaces
None

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

3.4

Communication Interfaces
None

Company Confidential

Page 7 of 22

Product Requirement Specification for Mobile SIP Dialer V1.0

4. Features & Functional Requirements

4.1 Support Device (Priority 1)
The followings are the platform to be supported 1. Nokia Phone a. Symbian 2nd Edition feature pack 1, 2nd Edition feature pack 1 and 2nd Edition feature pack 2 We do not have 2nd edition and we never intended to have it as our roadmap because we think 2nd edition phones are rarely used and have other issues also. (sin ming ok) b. Symbian 3rd Edition, 3rd Edition feature pack 1 and 3rd Edition feature 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 shall replace the voip call. The client shall support RFC 3621.

4.2

Mobile Client Features (Priority 1)

4.2.1 Making and receiving VoIP calls (Priority 1) The client shall be able to make to 1. PSTN number (Off net call) OK
Company Confidential
Page 8 of 22

Product Requirement Specification for Mobile SIP Dialer V1.0

2. 3.

Another client (On net call) OK 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)OK 2. 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 3G and 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 the 14kbps low bandwidth of GPRS.
The client shall detect that the user is using the GPRS network and adopt a different strategy to accommodate the low up ramp bandwidth of the GPRS network. The client shall pack 4audio frame in a single audio packet send over Companys 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 same technique even in WIFI network or 3G? (sin ming. The strategy for compacting multiple frame in 1 audio packet can degrade audio quality. Therefore I would like to use it only for GPRS and 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 make VoIP 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 the client.
The user shall be able to set the number of call detail he wants to view.
Company Confidential
Page 9 of 22

Product Requirement Specification for Mobile SIP Dialer V1.0

(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 mode and back to handset mode. 4.2.8 Configurable Seamless trigger of Company VoIP call. (Priority

1)

Mobile Client shall run in the background. It will automatically comes to the 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 registered with sip proxy, the client will automatically makes a voip call. If not registered, 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 or VoIP 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 VoIP Call shall be made. If the client is not registered to Company VoIP, the client shall automatically make a GSM call without asking. If we are deciding between VOIP or GSM call on the basis of the fact if dialer is registered or not what is the point of asking from user? Please explain where this dialog will fit. (sin ming. The dialer shall only prompt the user if it is registered with our sip proxy. The user might choose to make a GSM call if he is calling a local call. If the dialer is not register, there is no point asking since it cannot make a 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 shall make a normal GSM call. The client shall be able to intercept call make from 1. The native phone call home menu 2. The native phone phonebook 3. The native phone call record
Company Confidential
Page 10 of 22

Product Requirement Specification for Mobile SIP Dialer V1.0

a. Dailed number b. Received number c. 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. OK 1. 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 that there is a new client. The user can still use the service even if he did not 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
Company Confidential
Page 11 of 22

Product Requirement Specification for Mobile SIP Dialer V1.0

suggest we use only one proxy and this proxy we get from one single http response which was used to tell us if upgrade is available at startup. From backend you change proxy whenever its down. Suggest me (sin ming, The sip proxy failover is necessary to avoid service down time 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 return by 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 the client.
The sip proxy shall return the account balance in a proprietary header in the 200 OK message (registration response). OK so there will not any http URL called to fetch balance? (Sin Ming. The account balance is currently 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 header in 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 increment every seconds. The live timer shall be displayed in the following format hh: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 on these number to redial them.last 5 dialed numbers? Do you mean call history (which comes via your API?) or nokia phones dialed number list. According to functionality we will capture calls from contact, local nokia call log etc anyway. Please explain. (sin ming Can you put the voip call made from the dialer into the nokia phone dialed list? Basically our symbian 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
Company Confidential
Page 12 of 22

Product Requirement Specification for Mobile SIP Dialer V1.0

combined this feature with the 4.2.20 VoIP Call History. Basically allows the 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 less than a specified amount.
The SIP Proxy shall send the alert in the 200 OK message in response to the client registration message.OK full format needed. (sin ming. We will 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 voip call.
The client should be able to switch between the active call and call which is on hold. If the user is engage in a gsm call and a call voip call came in, in which the 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 symbian disconnects GPRS anyway. We will try to see what control can have to identify such event. For 3G and WIFI hold can be implemented but we will decide the exact functionality 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 and allowed 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 order 1. G729 2. G723 not available (sin ming understood) 3. iLBC 4. GSM 5. AMR
The client shall as far as possible use codec that is supported by the device and avoid using soft codec. OK presently we are using device specific codecs only. In some devices iLBC and g729 are there (Like symbian 3rd edition and 5th edition) but in most of the devices g729 is not there. 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 we can opt for third party hardware optimized codecs for the platforms where such codecs are avialble. GIPS, Voiceage and SPIRIT DSP are options.
Company Confidential
Page 13 of 22

Product Requirement Specification for Mobile SIP Dialer V1.0

4.2.20 VoIP Call History (Priority 2) The client shall keep a record of all the voip calls made, received and missed by the client. These call records shall be view in the client and also be inserted into the native phone call history (if possible). OK I think for all kinds of call we will get output from http APIs? (sin ming. Actually I am thinking of getting the phone to keep a record of the dialed number, received call and missed call. We use the http api for the 4.2.6 to get the duration and charge for the call. Again, we probably can combine 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 if the user picks up the call in another device, that call should not be marked 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 the message which is to be send to the caller. Please explain this feature. I think you already have a voice mail box. It would be good if Do not disturb flag can be set in your server and from there itself it can take to voicemail. In many platforms implementing voice message (Local ) would be an issue, if it is server based it will be for all platforms . Please comment (sin ming after consideration, we decide to remove this feature) 4.2.22 Caller ID Display (Priority 1) The client shall be able to display the caller ID when the client receives an on net call. OK. To be more clear. In every incoming call Invite we will have caller id like Manoj <sip:1234@Company.com> where Manoj 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 the user user ID (MRTalk ID) and password.OK
The client shall also ask the user if he wants to auto start the client whenever the phone starts up.OK If the user selects yes, the client shall be started whenever the phone restarts.OK The user can change this selection through the client settings.OK
Company Confidential
Page 14 of 22

Product Requirement Specification for Mobile SIP Dialer V1.0

4.3

Serving Advertisement when user make call OK (Priority 1)

The client shall display advertisement to the user before and after the call. The advertisement display shall be based on the user profile captured 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 audio advertisement to play. 3. The client open a browser within the client and display the banner 1 URL. 4. The client shall then send an sip invite message to the sip proxy, passing to it the audio advertisement to pay in the proprietary header. 5. The sip proxy shall play the audio advertisement and then connects the call. 6. When the call is ended, the client shall display another banner 2 within the embedded browser. 7. Note that the client shall provide click through for both the banner. In another words, the client shall open the browser and direct the browser 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, the ad server will not return any banner. Under this circumstances, the client need not embedded the browser in the dialer).

4.4

Fund Transfer (Must have) OK (Priority 1)

The client shall have the user interface for performing fund transfer. The user can 1. Transfer to Another Company Account For this feature, the user can transfer balance from his account in the client to another account specified. The user can specify the amount he wants to transfer.
Company Confidential
Page 15 of 22

Product Requirement Specification for Mobile SIP Dialer V1.0

2.

Fund transfer to 3rd party.

3. Recharge The 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 fund transfer function.)

4.5

CallBack (Priority 1)
The client shall be able to set the phone to make a voip call or a callback call. 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 always make 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 callback server will disconnect incoming call (Not Company client) (sin ming yes. Sorry about the mistake) d. The client shall auto pick up the call and automatically send the dtmf of the destination he wants to call. e. Company callback proxy shall then connects the client to the destination. (Detail design for this service shall be provided by Company) Please provide detailed design (sin ming. We are currently working on it. Will provide the detail in the earliest possible moment)

4.6

SMS (Priority 1)
The client shall be able to send sms. The client shall have a editor for the user to enters his select the destination 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
Company Confidential
Page 16 of 22

Product Requirement Specification for Mobile SIP Dialer V1.0

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

4.7

Anti-blocking mechanism (Priority 1)

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

4.8

Presence/IM (Priority 2)
The client shall have all the Instant Messages features (IM) such as 1. Presence 2. Click on user to a. Send sms b. Chat c. Make call 3. 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 be implemented. Implement Simple supported server or implement Simple in your SIP proxy itself (instead of directly accepting XMMP messages from client). This way Company client to client Voice also supported using a SIP message. 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 talk independently to jabber server in case user selected IM however in case of voice call SIP proxy will be used. Let me know what you think about it. After I get more information about the kind of network you are running I can assist more. If you already have some information, For voice call between Company client and yahoo/gtalk or MSN, please share with us to speedup.

4.9

Voice Mail (Priority 1)

The client shall be notify of any voicemail for him (using message waiting indicator MWI). This message list will be fetched only at application startup or after certain intervals? The user can click to access his voicemail.
Company Confidential
Page 17 of 22

Product Requirement Specification for Mobile SIP Dialer V1.0

(Details shall be provided by Company on the voicemail interface) OK more questions may be there after we receive details.
(sin ming. Our current system uses subscribe and notify to get the voicemail Alert.The flow is briefly describe as follow) The sip proxy shall inform the client the voicemail server IP for this client in the registration 200 OK message. The client will then send a sip subscribe message to the 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 the voicemail. The client shall display on the UI that a voicemail is present. When the user clicks to listen to the voicemail, the client shall make a VoIP call to a certain number (to the voice server) with the voicemail server IP in the Route field.)

4.10 Call Forward (Priority 1)

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 call forward number) I think some screen could be provided in Settings to set forward number. I guess you will provide some http API to set it to your server? (sin ming. Yes, we will)

4.11 Call Transfer (Priority 1)

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 call transfer)
I think this would be a difficult thing to achieve as in most of the platforms we plan use device specific APIs for all audio handling for better performance. To switch between calls would require custom audio mixer. But if we decide to use third party media engine like SPIRIT DSP we can do that easily I think. Please comment.

4.12 Conferences (Priority 1)

The user shall be click to participate in a multiply parties conference by calling to Company conference server. The conference server shall return in a proprietary header 1. Status of the conference 2. 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
Company Confidential
Page 18 of 22

Product Requirement Specification for Mobile SIP Dialer V1.0

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

4.13 Centralized Phone book(Priority 2)

The client shall be able to load all his contacts from his native phone to the 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 the contacts. (Details on the centralized phone book interface shall be provided by Company). OK we will wait for the details.

4.14 Multiple Language Support (Priority 2)

The client shall support multiple language. Some of the language to be supported are 1. English 2. Mandarin 3. Hindi 4. Malayalam 5. Tamil 6. Arabic The user select the language on the client . When set, all the text in the client shall be display in the language selected. The IVR shall also be display in the language selected. The client shall send the user selected language to the sip proxy which will instruct the IVR to play the language selected. (Details on the provided later). OK

4.15 Value Add Service Content (Priority 1)

Company is currently working with spice to provide content such as 1. Music 2. Chanting 3. Sport etc The user shall be able to access the content easily from the mobile client. Spice Content Server shall provide the list of contents (this list can change dynamically).
Company Confidential
Page 19 of 22

Product Requirement Specification for Mobile SIP Dialer V1.0

The client shall be able to interface with spice content provider server to get 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 mp3 format) and play it using the clients native media player. Details shall be provided by Company. More details required for this to comment.

4.16 Interop with Other Client (Priority 2)

The client shall be able to interwork with other major IM client both in terms of voice and IM functions. IM such as 1. Yahoo 2. Skype 3. MSN 4. Gtalk The interop is done at the Company backend. The client must provide an interface for the user to specify the IM type which 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)

4.17 Fault Reporting from Dialier (Priority 1)

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 option 2. 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 to Company fault report server via a http api. The type of information to be collected and the httpd api shall be define by Company.

When we say fault do we mean Call failure reasons? if yes do you mean that on every call failure Not found, Temporary failure we have to ask user to submit information? (sin ming. We have not exactly define the nature of the report yet. But at a minimum, I need to know the username, the destination 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)

Company Confidential

Page 20 of 22

Product Requirement Specification for Mobile SIP Dialer V1.0

4.18 Contact Us Email (Priority 1)

The clients shall have Company Contact Us email in which the user can access to easily. Basically the user just click the Contact Us options and Companys support center email is presented to him.
If you can provide me an http URL which can save some content directly in your ticketing system, it would 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).

5. Reporting and Monitoring Requirements 6. Other Nonfunctional Requirements

6.1 Performance Requirements
Call Setup Within 10 seconds Able to handle 2000 registration Able to handle 500 concurrent calls

6.2 6.3 6.4

Safety Requirements
None

Security Requirements
Customers information must be kept secure.

Quality Requirements
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 care engineer.

6.5 6.6 6.7 6.8

Business Rules
TBD

User Documentation
System Installation Guide

Software statutory & regulatory requirement

None

Is a Trial version required?

Yes (Delete as applicable)
(If YES) N0

Company Confidential

Page 21 of 22

Product Requirement Specification for Mobile SIP Dialer V1.0

Trial Period:

6.9

3 rd Party product Compatibility List

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

Company Confidential

Page 22 of 22