1. Corrado

    Flashphoner as server for internal calls

    We need to talk from a webpage to a raspberry, would it be possible to do it using webcallserver? Maybe using it as an internal gateway without a sip server. If so is there any docs around? thanks, Corrado
  2. A

    Mixing of SIP calls and WebRTC calls in Conference

    Hi there, Is there a possibility of Mixing of SIP calls and WebRTC calls in Conference? kindly assist... Regards AB
  3. Alex YYY

    Проблемы с docker контейнером

    Имеется docker-compose файл с такими настройками: wcsserver: image: flashphoner/webcallserver:5.2.898 container_name: wcsserver network_mode: host expose: - "8080-8084" - "8443-8445" - "554" - "1935" - "1935/udp" - "30000-35000"...
  4. S

    При SIP звонке слышно только одного собеседника

    Тестировали на Есть номер A и номер B. Если совершить звонок через браузер, то звонок идет все корректно, но браузер звук не воспроизводит(тестировали во всех браузерах), в тоже время, собеседник слышит то что говорится через браузер. Тестировали SIP...

    Browser Web Phone with SIP IP phones and Mobile GSM

    Dear experts, i followed the following link,
  6. N

    Звонок без sip сервера

    Добрый вечер! Подскажите, возможно ли организовать аудио звонок с браузер на браузер?
  7. T

    Через 30-40 секунд звонок отключается

    Здравствуйте. У меня есть проблема при звонке. Звонок отрывается через 30-40 секунд. Я использую СИПы из платформы
  8. M

    Delayed calls without human action

    Hi, we are receiving some complaints from our clients that assure that their apps are making calls without any action. We are using the webcallserver as gateway from the webrtc app in android to SIP. We initially thought that the problem was in the mobile app, but we finally reproduced it, and...
  9. A

    SIP connectivity with opensips

    Hi there We are trying to make a connectivity with SIP API, we have successfully installed opensips on WCS and its running, kindly let us know how to define the sip extension on this and how to connect the web client? Thanks AB
  10. S

    call browser from mobile

    Hi, I saw this article: I was wondering whether the opposite is possible as well. Can I call a browser from a mobile phone? best regards Piotr
  11. vanarie

    Twilio group sceen to RTMP

    Hello, I've been testing with WCS and setup a test server. I was able to get webRTC -> WCS -> RTMP working from the install and read the rtmp://myserver stream with VLC media player. I assume that's a good test. My goal now is to find a good, scalable way to port a video meeting conference...
  12. A

    Webphonr to Sip phone connectivity

    Hi there We would like to connect peer to peer connectivity with the webphone to a sip phone and vice versa the clint sip phone is running on a LAN where WCS is running... The local sip phone is xlite and does not require any registration or user name... Kindly suggest what information we need...
  13. L

    REST - Starting call with hasAudio false still uses sendrecv in SDP causing call to timeout

    Using Web Call Server Version 5.2.755 I'm passing the following to /rest-api/call/startup { "callId": "1234567", "callee": "1234567", "hasVideo": false, "hasAudio": false, "sipLogin": "[login]", "sipAuthenticationName": "[authName]", "sipPassword": "[password]"...
  14. Dolph

    Web Call server error

    Hi, I'm experiencing the Web Call Server, after I've installed the server and go to the web interface, click the "Phone" on the left nav list and reach the "Phone Min" page. I filled out all the SIP server information and click "Connect", it shows "ESTABLISHED" , but then I filled out the number...
  15. K

    Does WCS3 support latest chrome/firefox browser

    Hi All We have an application built with Flashphoner WCS3(FlashphonerWebCallServer-3.0.1721) + Asterisk.. Call signalling work prefect but there is no audio after answer call. I also test with WCS demo client(WCS-client-3.0.519) Phone.html, PhoneJS.html, and Click2callJS.html with latest...
  16. B


    Hi, After reading the documentation I can't find a way to use WCS as RTMP gw and convert RTMP (input) to SIP/RTP (output) , only SIP/RTP to RTMP. Is it possible? Is there an exemple? Thx in advance
  17. M

    Calls interrupted in iOS when receiving incoming in-app notifications or when the screen saver is on

    Hi, we are receiving complaints from our customers since they are having some failures during a live call from iOS to a normal phone (we use WCS as webrtc/SIP gateway). To be exact, the call is estalished but when the user receives an incoming inapp notifciation (e.g whatsapp) the call is...
  18. M

    Click to call attacks prevention

    Hi, we are developing a click-to-call webapp for one of our customers. We are worried about potential attacks to our systems and we don't want to include a captcha control before the click to call (avoiding bots). Does WCS support any kind of protection against these kind of attacks? We want to...