Softphones srtp trabalhos
...Android customised to my company with the following features. Main features : Audio call Instant Messaging Pictures and files sharing Account creation assistant Address Book Languages : English, French Call History Display of advanced call statistics Echo Cancellation Quality of Service Secure communications : zRTP (available only under GNU GPL v2 license), TLS, SRTP Advanced features : Audio codecs : G711, SPEEX, G722, AMR-WB (G722.2), GSM 6.10, AMR-NB, ILBC, SILK, G729 Integration with push notification (requires compatible SIP server) Nat friendly: guesses NAT address for SIP messages, uses STUN for RTP streams. Integrated ICE support (RFC5246) to allow peer to peer audio & video connections without media relay server. Low bandwidth mode: make audio
We have our own telephony infrastructure and we now require a SIP softphone App to connect to our platform for our team to use. I need a sip softphone application for Android and IOS, with support for TLS,SRTP,ZRTP and XMMP(encrypted). Features required. * Full support for TLS,SRTP,ZRTP and XMMP(encrypted). * Full Source Code to make mods after project completion. * Support G729, G723, Codecs * Application should work all Networks 3G/Wi-Fi/2G or GPRS. * Phone full contact book sync * Real time Balance Display * Loudspeaker * Account to Account Balance transfer * Call Duration * Call Log * Redial/Call from Call logs * Simple to use User Friendly Interface. I need only expert and very experienced in SIP Phones to integrate with our existent FreeSwitch+A2Billi...
Hello We have our own telephony infrastructure and we now require a SIP softphone App to connect to our platform for our team to use. I need a sip softphone application for Android and IOS, with support for TLS,SRTP,ZRTP and XMMP(encrypted). Features required. * Full support for TLS,SRTP,ZRTP and XMMP(encrypted). * Full Source Code to make mods after project completion. * Support G729, G723, Codecs * Application should work all Networks 3G/Wi-Fi/2G or GPRS. * Phone full contact book sync * Real time Balance Display * Loudspeaker * Account to Account Balance transfer * Call Duration * Call Log * Redial/Call from Call logs * Simple to use User Friendly Interface. I need only expert and very experienced in SIP Phones to integrate with our existent FreeSwitch...
I own a Synology nas on which I installed the asterisk package. It works correctly, but I have to add a sip voip service that requires SRTP and tls but are not currently installed. I am looking for someone who can install it via teamview. No compilers installed yet
I own a Synology nas on which I installed the asterisk package. It works correctly, but I have to add a sip voip service that requires SRTP and tls but are not currently installed. I am looking for someone who can install it via teamview. No compilers installed yet
...make calls (SIP protocol) to our call center for free. Actual programming in some cases is MINIMAL (android -> use opensource Csipsimple -> only change to have preconfigured SIP account in apk install file) We would like for open source software (sip phones) to be used to save time and money. On most platforms there are ready made softphones that fill our riqueirments and only a preconfiguration of a sip account needs to be made). Examples of softphones using PJSIP: Android -> we currently use this and are quite satisfied and it would need no modifications, just preconfigure a sip account. iPhone/iPad -> WebRTC - open source SIP http://webrtc2sip
I own a Synology nas on which I installed the asterisk package. It works correctly, but I have to add a sip voip service that requires SRTP and tls but are not currently installed. I am looking for someone who can install it via teamview. No compilers installed yet
...TURN, and ICE Plug-and-play voice codecs including G.711, G.729, GSM, iLBC, SirenTM, Speex, and more Voice processing (AEC, AGC, CNG, NS, VAD) for hands-free handsets and tablets Conferencing for up to 8 parties Peer-to-peer media transport over UDP-enabled networks for low latency and high scalability Scalable to tens of millions of subscribers Security: TLS for signalling encryption SRTP for media encryption YOU...
We have a VOIP Server and we want to launch our own Soft-phone app for Android and iPhone. The goal is to modify the existing open source app like CSipSimple ( ) or Linphone () to work with openvpn open source app. 1) Must be configurablle t...like CSipSimple ( ) or Linphone () to work with openvpn open source app. 1) Must be configurablle trough web conf file 2) Remove the Video capability (interested in only text + voice) 3) Change the UI settings (Color + Font) 4) Display our logo in the UI. 5) Must start at phone boot. 6) Password protect setings 7) Must use TLS SRTP security. 8) Must be reacheble 24/7 9) Test 10) Publish to Google Play and App Store.
Project Description: CSIPSIMple modifications 1. Our Icon and Company Name on the application 2. First launch no setting screen will be presented to the user but only the following: User name : Password : server : port : (Default 5060) transport : UDP / TCP / TLS (Default UDP) audio Protocol: RTP/SRTP/ZRTP (Default RTP) codec: G711 ulaw/alaw , G729, (Default ulaw) Save/edit/delete 3/ Background colour white 5/ For the following logging no setting screen unless explicitly called from the android menu to the above screen.
Implement a scalable VoIP Telephone System that will interconnect 3 colleges. - User in college A should be able to communicate with user in college B from their local networks (even if the users don't have internet). - IVR configuration is needed to cater for the college administrative needs. - A web interface to register the SIP users and Billing - Implement QoS (on the router) for V...with user in college B from their local networks (even if the users don't have internet). - IVR configuration is needed to cater for the college administrative needs. - A web interface to register the SIP users and Billing - Implement QoS (on the router) for VoIP within the college to prioritize voice. IMPORTANT NOTICE * You are only setting up for one college * The users are using on...
Implement a scalable VoIP Telephone System that will interconnect 3 colleges. - User in college A should be able to communicate with user in college B from their local networks (even if the users don't have internet). - IVR configuration is needed to cater for the college administrative needs. - A web interface to register the SIP users and Billing - Implement QoS (on the router) for VoI...user in college B from their local networks (even if the users don't have internet). - IVR configuration is needed to cater for the college administrative needs. - A web interface to register the SIP users and Billing - Implement QoS (on the router) for VoIP within the college to prioritize voice. IMPORTANT NOTICE * You are only setting up for one college * The users are using on on...
okay so we neeed someone to setup a auto or predective dialer system for our business . we were told that we could use for the softphones . skype . google voice . yate client ans we were told that al we need is internet connection. we are lookijg for someone to set this up for cheap and effecient. whatever softphone we use has to be an auto dialer or predctive we are looking for and soulution that is quick and easy note: we were told we didnt need a hosted soultion all we need is strong ineternet and a softphone solution we are open for suggestions thanks
...voice and SMS with instant messaging or chat, live video sharing and file transfer across all devices and networks Standalone Messaging 1-to-1 Chat Group Chat File Transfer Content Sharing Social Presence Information IP Voice call Best Effort Video call Geolocation Exchange Network based blacklist Capability Exchange based on Presence or SIP OPTIONS UDP, TLS, TCP, STUN, TURN, ICE, RTP, SRTP, Relay proxy Phase 1 : Server + iOS + Android = 3 weeks delivery Phase 2 : PC + Mac = 2 weeks delivery Phase 3 : Windows Phone + Kindle = 2 weeks delivery Phase 4 : BB + Nokia = 2 weeks delivery Payment will be subject to delivery of live app after each phase with satisfactory and completion test for all features inclusively. NO upfront or partial payment will be made. ...
I own a Synology server on which I installed the asterisk package. It works correctly, but I have to add a sip voip service that requires coding and SRTP tls but are not currently installed. I am looking for someone to install them.
Hola, somos una empresa pequeña de turismo receptivo en Bogotá y queremos implementar un sistema de PBX virtual que nos ayude con la recepción de llamadas y el manejo de estas. Los requerimientos son los siguientes. Requerimientos básicos 1....El usuario debe escuchar una grabación hasta que le atiendan la llamada. 7. En las opciones 1,2 y 4, en caso de no obtener rspuesta después de X segundos de espera, se desvía a un buzón de voz con notificación a un correo. -------------- Requerimientos Adicionales 1. Sería ideal que al contestar una llamada, nuestros agentes sepan las opciones que ha elegido el cliente. 2. Si los softphones instalados en smartphones de las opciones 1,2 y 4 no están ...
...TURN, and ICE Plug-and-play voice codecs including G.711, G.729, GSM, iLBC, SirenTM, Speex, and more Voice processing (AEC, AGC, CNG, NS, VAD) for hands-free handsets and tablets Conferencing for up to 8 parties Peer-to-peer media transport over UDP-enabled networks for low latency and high scalability Scalable to tens of millions of subscribers Security: TLS for signalling encryption SRTP for media encryption THANK YOU...
...TURN, and ICE Plug-and-play voice codecs including G.711, G.729, GSM, iLBC, SirenTM, Speex, and more Voice processing (AEC, AGC, CNG, NS, VAD) for hands-free handsets and tablets Conferencing for up to 8 parties Peer-to-peer media transport over UDP-enabled networks for low latency and high scalability Scalable to tens of millions of subscribers Security: TLS for signalling encryption SRTP for media encryption THANK YOU...
...TURN, and ICE Plug-and-play voice codecs including G.711, G.729, GSM, iLBC, SirenTM, Speex, and more Voice processing (AEC, AGC, CNG, NS, VAD) for hands-free handsets and tablets Conferencing for up to 8 parties Peer-to-peer media transport over UDP-enabled networks for low latency and high scalability Scalable to tens of millions of subscribers Security: TLS for signalling encryption SRTP for media encryption THANK YOU...
...TURN, and ICE Plug-and-play voice codecs including G.711, G.729, GSM, iLBC, SirenTM, Speex, and more Voice processing (AEC, AGC, CNG, NS, VAD) for hands-free handsets and tablets Conferencing for up to 8 parties Peer-to-peer media transport over UDP-enabled networks for low latency and high scalability Scalable to tens of millions of subscribers Security: TLS for signalling encryption SRTP for media encryption YOU...
Telemarketing to source for prospective clients. Do you think you speak well? Are you able to converse easily with strangers? Do you possess basic microsoft excel skills? Do you have experience working with any softphones (ie. zoipper)? If you do, this job suits you. I am looking for a short term telemarketer right now. If you think you have what it takes, look no further!
Have a look at my project regarding VOIP. Freeswitch 1.2.23 Configured as Media Server - Multitenant Configuration - SRTP and TLS configuration - Configuration for the following clients - Counterpath Bria () - Global IP Juggler () - Snom Hardware Telephones () - Voicemail to Email - SIP presence configured (alternatively configured on Kamailio) - SIP Instant Messaging (alternatively configured on Kamailio) Kamailio 4.1 Configured as SIP Frontend Server in front of Freeswitch - Multitenant Configuration - SRTP and TLS configuration - Configuration for the following clients - Counterpath Bria () - Global IP Juggler () - Snom Hardware Telephones (http://www
Have a look at my project regarding VOIP. Freeswitch 1.2.23 Configured as Media Server - Multitenant Configuration - SRTP and TLS configuration - Configuration for the following clients - Counterpath Bria () - Global IP Juggler () - Snom Hardware Telephones () - Voicemail to Email - SIP presence configured (alternatively configured on Kamailio) - SIP Instant Messaging (alternatively configured on Kamailio) Kamailio 4.1 Configured as SIP Frontend Server in front of Freeswitch - Multitenant Configuration - SRTP and TLS configuration - Configuration for the following clients - Counterpath Bria () - Global IP Juggler () - Snom Hardware Telephones (http://www
Please find requirement below: Freeswitch 1.2.23 Configured as Media Server - Multitenant Configuration - SRTP and TLS configuration - Configuration for the following clients - Counterpath Bria () - Global IP Juggler () - Snom Hardware Telephones () - Voicemail to Email - SIP presence configured (alternatively configured on Kamailio) - SIP Instant Messaging (alternatively configured on Kamailio) Kamailio 4.1 Configured as SIP Frontend Server in front of Freeswitch - Multitenant Configuration - SRTP and TLS configuration - Configuration for the following clients - Counterpath Bria () - Global IP Juggler () - Snom Hardware Telephones
...create must provide communications in three styles: - Regular SIP based telephone calls - SIP based PTT mode (sender and receiver in application) - Short message system (based on external XMPP server) Basic application requirements are: - Android Jelly Bean, custom cell phone design, screen size 4,3” with dedicated PTT button - VoIP Calls and PTT must complain SIP standard - VoIP Calls use SRTP - Addidtional directory Services from VoIP Server. Server not part of this work, interface descriptions are given - All communication will be over LTE or WIFI, adaptive codecs must be available - Adaptive GUI design, colored scheme, intuitive usage. Screen design must be included. - All source code work shall be delivered in well documented form. The complete Server S...
We want the users/clients on A2billing-Asterisk server to be able to receive calls on their softphones and their IP phones. Presently all client with SIP created for are able to make calls using the softphones and the IP phones but non is able to receive income calls either between each other or from inbound DIDs. So we need some one that can edit the configuration to be able to do the above. This is a production server that is life 24/7. Thanks,
...TURN, and ICE Plug-and-play voice codecs including G.711, G.729, GSM, iLBC, SirenTM, Speex, and more Voice processing (AEC, AGC, CNG, NS, VAD) for hands-free handsets and tablets Conferencing for up to 8 parties Peer-to-peer media transport over UDP-enabled networks for low latency and high scalability Scalable to tens of millions of subscribers Security: TLS for signalling encryption SRTP for media encryption...
I have Asterisk 11.8.1 running with FreePBX 2.0.11.32. I have the new WebRTC module installed but I am having issue with SRTP Packets. I receive the following errors when making calls: [2014-04-09 08:43:29] WARNING[18338][C-00000014] sip/sdp_crypto.c: Could not set SRTP policies [2014-04-09 08:43:29] WARNING[18338][C-00000014] sip/sdp_crypto.c: Could not set SRTP policies [2014-04-09 08:43:29] WARNING[18338][C-00000014] chan_sip.c: Rejecting secure audio stream without encryption details: audio 25176 RTP/SAVPF 111 103 104 0 8 106 105 13 126 I need this asap
Is it possible to integrate the CRM Vtiger with the PBX service Call Centric and one of the following softphones Zoiper or X-Lite. If so I would like it set up ASAP. PBX PROVIDER: DIALERS: or CALL CENTRE CRM: That is the ideal set up I'm looking for. Unless you can think of a better CRM PBX ...
Im recently starting my first Asterisk project. After install it, everything is already configured and working properly thru softphones (X-Lite), but Im completely stuck how to configure SIP phones as Polycom IP650 with my Asterisk. Looking for somebody to help me configure and learn how to configure those phones now on. I have a small IT company in Miami, starting installing Asterisk servers to some clients. Recien estoy empezando a instalar mi primer servidor Asterisk. Todo esta funcionando bien con los telefonos X-Lite, pero no conozco nada de LINUX para poder configurar los archivos, el TFTP Server y demas cosas que segun veo se necesitan para conectar telefonos duros SIP. Quisiera ayuda para configurar un ejemplo y aprender a como configurarlos en lo adelante. Tengo una peque&...
Planowana aplikacja ma pełnić funkcję softphone (odpowiednio obrendowanego) dla firmy telekomunikacyjnej. Aplikacja musi być oparta o protokół SIP - (SIP/UDP, SIP/TCP, SIP/TLS) Aplikacja musi wspierać IPv6 Aplikacja musi rozpoznawać kody klawiszy DTMF Dodatkowo wymaganiem opcjonalnym jest to, aby aplikacja pozwalała zarejestrować si...połączenia sumultaniczne - tj. zawiesić jedno połączenie i wykonać następne Obsługa SIP proxy (register, proxies, outbound proxies) Aplikacja powinna obsługiwać "text instant messaging" z notyfikacją dostarczenia Obsługa DTMF zgodnie ze standardem RFC2833 oraz SIP INFO Aplikacja musi wspierać popularne kodeki: np. speex, G711 alaw i ulaw, GSM, G722, G729 (wymagana licencja) iLBC. Obsługa SRTP, zRTP Obsługa automatycznego usuwania ech...
Planowana aplikacja ma pełnić funkcję softphone (odpowiednio obrendowanego) dla firmy telekomunikacyjnej. Aplikacja musi być oparta o protokół SIP - (SIP/UDP, SIP/TCP, SIP/TLS) Aplikacja musi wspierać IPv6 Aplikacja musi rozpoznawać kody klawiszy DTMF Dodatkowo wymaganiem opcjonalnym jest to, aby aplikacja pozwalała zarejes...na połączenia sumultaniczne - tj. zawiesić jedno połączenie i wykonać następne Obsługa SIP proxy (register, proxies, outbound proxies) Aplikacja powinna obsługiwać „text instant messaging” z notyfikacją dostarczenia Obsługa DTMF zgodnie ze standardem RFC2833 oraz SIP INFO Aplikacja musi wspierać popularne kodeki: np. speex, G711 alaw i ulaw, GSM, G722, G729 (wymagana licencja) iLBC. Obsługa SRTP, zRTP Obsługa automatycznego usu...
...==> Cable modem ==> DDWRT router ==> my computers I would like to somehow use my connection to provide external access to three different servers located in my home. I will have the following servers: 1. Vicibox/Vicidial server (based on Asterisk) where users can use their softphones to register to. Will have its own static IP. 2. A separate Asterisk server where users can use their softphones to register to. Will have its own static IP. 3. An OpenSIPs/Asterisk server where users can use their softphones to register to OpenSIPs. Will have its own static IP. My issue is I am unsure whether I need 4 separate internet connections to make this work (one for personal, and one for each of the servers). Hopefully there is a way I can do some sor...
...NOT bid if you are not 110% confident you ALREADY KNOW how to do this. Your bid MUST INCLUDE the # of hours you expect this to take. In addition to the video, some other bullet points are: - For calls that don’t pick up after X seconds, disconnected #s, busy tones, the system should just skip over those and mark those as called. - Need to be able to set caller ID for each of the softphones registered - Multiple simultaneous users may be in sessions simultaneously (user 101 shown in the example, but could also have folders for 102, 103,...)...
I need you to create the ssl certificates and setup freepbx and aastra and yealink ip phones so that they will communicate using TLS for sip and SRTP for rtp packages. Only bid if you have experience with an identical scenario. MAX budget is $40! Please do not place higher bids if you want to be considered. Thanks
We are looking for a person with experience and knowledge in deploying VoIP solutions. 1) This project involves deploying a Multi-tenant PBX solution. 2) Kamailio will Load balance traffic between the FreeSWITCH servers located in different locations. 3) Managing users will be done through a central PostgreSQL Replicated DB 4) Users will connect via TLS to Kamailio / SRTP for the media Please feel free to contact me for further Questions. Thanks! David
...DTMF: the ability to enter numbers to use an auto attendant Via RFC 2833, SIP INFO and in-band Advanced Features Network Traversal Application Managed, Server Managed or Custom Configuration Global IP Support STUN & ICE Media Efficiency and Quality Noise Reduction Echo Cancellation Voice Activity Detection Quality of Service Codecs Variable Bitrate Security and Encryption TLS and SRTP (Secure call signaling and audio encryption) DNS SRV record lookups Call quality statistics VPN support...
We are looking for a Voip Pro that can setup a Freeswitch cluster using FusionPBX. Requirements: 1. Quick delivery - before Christmas. 2. SIP TLS + SRTP. 3. Centrally managed DB for both freeswitches 4. Presence enabled. 5. Documentation & knowledge transfer Attached file. (Please see SETUP A)
We are looking for a person with experience and knowledge in deploying VoIP solutions. The idea is to create a sip Loadbalancer that handles the sip traffic and routes the RTP to Freeswitch servers located in multiple sites. the system needs to support SRTP and hold a central DB for the directory Users and Dialplans. Please feel free to contact me for further Questions. Thanks ! David
Implementing advanced Asterisk clusters including load balancing, TLS, SRTP, Encrypting the Sip trunks between servers in Amazon Cloud environment. In later stage we will be doing Opensips implementation to back end with Asterisk requiring knowledge of dial plans, load balancing, rtp media proxy.
I need a SIP dialer application for iOS and Android with the following features. - Can use any opensource SIP SDK for developement (Linphone can be used as open source) - Support fo...application for iOS and Android with the following features. - Can use any opensource SIP SDK for developement (Linphone can be used as open source) - Support for G729, SILK, and G711u - Should work with any standard SIP Soft Switch (Asterisk) - Soft Switch's IP Address should be embedded in the Code. - User should be prompted to give in Username and Password Only. - Should support TLS/SRTP - I need the dialer to work with A2Billing. - Account Balance and Calling destination should be displayed on the screen upon dialing the number. - Should have the functionality of working in b...
SIP-based phone for Apple IOS (Android, Blackberry, Windows Mobile) that uses a Wi-Fi or cellular data network connection to make and receive calls. Similar to Counterpart Bria or Acrobits Softphone, but customized. Feature...for Apple IOS (Android, Blackberry, Windows Mobile) that uses a Wi-Fi or cellular data network connection to make and receive calls. Similar to Counterpart Bria or Acrobits Softphone, but customized. Features and support for: - standard protocols SIP(S) - Audio codecs: G.711 a/u, G.722 (HD), iLBC, GSM, OPUS and SILK - Network Traversal - Security and Encryption (AES128/AES256), TLS and SRTP (Secure call signaling and audio encryption), ZRTP - VPN support - Remote management/configuration - Encrypted messaging Compatible with: - Asterisk...
...the telemarketing team). if someone is calling that number, then it should get routed to PBX where the incoming call can get routed by pressing 101, 102, 103 to go to individual employees. ( 5 employees with Skype number 347-xxx-xxx and 5 with the Skype Connect Sip account number. Only these last employees will use Vtiger for now (only Telemarketing team (5 Employees).) ).Employees have got a softphones installed on their computers (X-lite3). We need this virtual PBX to do the following: a. 718-xxx-xxxx is our company number. if someone calls that number, then it will prompt for 101 to employee 1 , 102 to employee 2 etc. 100 for operator etc. ( Same for the Skype Connect Number (Voip number), for the Telemarketing team) b. We eventually will also add more employees. c. Some ...
Looking for someone to help guide me through a few questions with SIP phones. This could end up being a longer term relationship as we will be using VoIP more over the next couple months. My current issue today is wanting to transfer two numbers so that incoming calls to those numbers go to softphones on computers here in the office. What do we need to make this happen?
Hello! I am looking for an individual to help setup an IVR on my Asterisk PBX. It is connect using the Google Voice Modif and I already have some extension setup, but right now I can only route the inbound call to...and it doesn't start the IVR system recording that I made from a text to speech to MP3 program. I need it to say. "Welcome to CentricStores, If you know your parties extension, please enter it now, otherwise for sales press 1, for customer service press 2," and so on... I don't have a super big budget, and this is connected on my home office. I plan on getting regular ip sip phones instead of softphones shortly, but I expect to have about 10 phones on this system. I have read forums about people getting this to work, by routing the Google Voice number...
...call quality Support for DTMF: the ability to enter numbers to use an auto attendant Via RFC 2833, SIP INFO and in-band Advanced Features Network Traversal Application Managed, Server Managed or Custom Configuration Global IP Support STUN & ICE Media Efficiency and Quality Noise Reduction Echo Cancellation Voice Activity Detection Quality of Service Security and Encryption TLS and SRTP (Secure call signaling and audio encryption) DNS SRV record lookups Call quality statistics VPN support Accessories Headset with microphone Headphones (no microphone): Bluetooth™ support: dependent on Android device and operating system...
...(most produce 150 + sales per per month) payment is paid weekly by bank wire, paypal or western union we will pay as you go as you meet our agreed bench marks. We have call centers making alot of money with this ongoing project and you should make alot also. Scripts, Rebuttals, Product Info provided. rnWE will also provide you with all live incoming leads and voip softphones. It is important that all agents must speak Fluent English. rnrnView our exciting product video to understand what the system does @ rnVisit our website @ rnrnThank You. rnrnGary Parker
...are looking for call centers that can produce 30-50 sales a month, payment is paid weekly by bank wire, paypal or western union we will pay as you go as you meet our agreed bench marks. We have call centers making a lot of money with this ongoing project and you could make a lot also. Scripts, Rebuttals, Product Info provided. We will also provide you with all live incoming leads and voip softphones. It is important that all agents must speak Fluent English. Thank You Maria - Director About Us: We are an Australian company who growing day by day. Most of our dealings are online and we have new products launching every quarter. This new venture has a goal is to reach 500 listings by the end of the year. The basic listings are $250, and you will earn $30-$50...
...looking for call centers that can produce 30-50 sales a month, payment is paid weekly by bank wire, paypal or western union we will pay as you go as you meet our agreed bench marks. We have call centers making a lot of money with this ongoing project and you could make a lot also. Scripts, Rebuttals, Product Info provided. We will also provide you with all live incoming leads and voip softphones. It is important that all agents must speak Fluent English. Thank You Maria - Director About Us: We are an Australian company who growing day by day. Most of our dealings are online and we have new products launching every quarter. This new venture has a goal is to reach 500 listings by the end of the year. The basic listings are $250, and you will earn $30-$50...
The project is creating a softphone that supports: direct IP-to-IP calls, phonebook, SIP,SRTP. works on windows and android.
