10178 Skypecopy standalone 4 website
N/A
Paid on delivery
EXPERTS ONLY MUST HAVE A SAMPLE I CAN VIEW ONLINE AND TEST BEFORE YOU GET PAID! I am looking for a skype standalone application. That I can be easily integrated in to my web site designed in PHP. PC TO PC first Pots later It will need to be integrated into existing user accounts and have all features of skype. When online notified of calls and directed back to [url removed, login to view] next button will be (Phone)opoly where a VOIP system will be integrated for members use. Here members of the site can make telephone calls to other members or make out side telephone calls to ?hard? phone lines. Users will have the options to assign their own ring tones, text chat, and smiley comments in chat plus many other cool features. This is a general overview, some minor details may change when the final coding is done. This is a general format of the Spaceopoly concept ADDITIONAL VOIP INFORMATION We are not merely using an open source software and giving it to you. Infant we are fully designing the client part of the software. e.g. the link you gave us, was the client side application, e.g. the application called "Skype", what the user sees and uses is the client side, and what is running at the back-end to connect the calls is the server side. We shall be designing the whole client side architecture, and even the server side, we are just taking parts from open source, so that we do not have to re-invent the wheel. "The SIP protocols shall be taken from Asterisk open source PBX, this would work on the server side asterisk Our price includes integrating the client with the website, i.e. it wont just be an application like skype, where the user has to register through the client. What you need is that when a user registers on your site, for using other services, he gets automatically registered with the VOIP client, and is able to use the same user id and password for all the functions shall as photo, video sharing upload, text/video chat and VOIP. Our program wont just be a standalone application like other VOIP software, in a way you can term it as a web-based VOIP system. Much more complex and advanced than other VOIP applications available, whether commercial or open-source. We are charging the amount not just for a standalone program, but integration into the site, to design a bridge between the VOIP application and the website. Javascript(' hotma') To explain, how the PC to POTS shall work. When a user logs in through the VOIP program, he shall get connected to your server (rackspace) . This is the server which has the interface cards installed, and telephone lines connected to them. So when a user wants to call from say USA to Germany, then first he shall be connected to your server via our voip protocols defined in the application, then if he chooses to talk to another user using the voip application then the server shall simply connect the two VOIP protocols to each other, and the users shall be able to talk to each other. But then when the user from USA chooses to call a normal telephone in Germany, then first he shall be connected to your server via our protocols. Then his call shall be routed to the interface card installed on your server. This will in turn connect the call to the no. in Germany through a local phone line. Then the user using our VOIP program shall be able to talk to the person in Germany using POTS. This means, that actually the user in USA will think that he has just made a long distance international call, and you can charge him the rates you want for it. But actually, it is just a local call for your server, i.e. you are paying the local rates for the calls. Now the responsibility of the website and the VOIP application designed by us will be to handle these services. Automatically after the user has finished calling, he shall be billed through the site. The logs shall be updated on the site, and if he has a deposit on your site, it shall be reduced. That is, we shall be making this applications so that it connects with the site. The following attachments or information should be included in this Statement of Work: VOIP Functional & Technical Designs SIP and STUN support [url removed, login to view] and ALSA (since 0.3) audio devices support G711u, G711a PCM, and GSM (since 0.3) codecs Audio codecs: G711 (alaw and ulaw), GSM ADPCM ,[url removed, login to view] (A-Law & X?-Law) GSM,Speex Audio drivers: OSS and ALSA (using Port Audio, a portable [url removed, login to view] API) Send DTMF (Dual Tone Multi Frequency) as SIP INFO Audio transport layer rtp Signalisation protocol SIP 2.0 rfc 3261) Management of multiple lines Handle of firewall symmmetric, restrictive port and address with STUN Support name display, Hold, blind Transfer, Mute, volume control. Caller Id(if the user has a no. otherwise his user id shall be displayed) Call Waiting. Call Conferencing(Upto 6 way conferencing on all voice calls) Call Logs(just like skype)Customized ringtones(as we decided) Ability to choose Online status(Like busy, appear offline, away mode, etc. just like in msn messenger and skype) Fully integrated Voice mail(ability to download messages directly on desktop, automatically) Caller id block( lets you block your own caller information when making a phone call. ) Speed Dialing( program dialing shortcuts of frequently called numbers to cut your phone number down to single digit dialing) Fully integrated Sms Application(as decided- lets you send smses thoughout the world to any no., the from no. will bear you user id or you VOIP telephone no.) Ringtones Exchange, Download and Upload feature. Online Connection Tree(displaying who is connected to whom) Fully integrated Video Chatting and Conferencing. Dedicated User phone numbers.(assigned by the admin,Also each user shall have his own id assingned.. for eg. 'X' will have #09HF56, then if you want you can assing a number to him say - 097848048, he can give this no. to his friends who can call him on this no.) Free Inter-calling. Personal Section included in user profile: Allowing user to create, a page to be viewed privately amonf friends or for public. The user can create a picture gallery also(as explained). Admin Controls- Ability to ban members. Ability to edit members info. Ability to assign dedicated phone numbers to members. Mass mailing feature(lets you send emails, newletters, files) Ability to connect to AIM, ICQ, MSN, Yahoo Messenger, and IRC. It provides capabilities not possible with original network clients, while supporting standard features such as audio chat, file transfers, group chats, chat rooms, buddy icons, multiple simultaneous connections to the same network, server-side contact importing, typing notification, direct connection (AIM), proxy support, encrypted messaging (AIM/ICQ), SMS support, and privacy settings.
Project ID: #1761043
