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  1. Max

    <iframe...mediaProviders=WebRTC,Flash,MSE,WSPlayer &autoplay=true&mute=1...>

    Embed Player sample application have no mute option, but Player sample can be simple modified to mute on playback start. Please look at WCS_HOME/client2/examples/demo/streaming/player/player.js script
  2. Max

    Chrome auto-play on mobile

    Hello. We tested Embed Player sample application with autoplay=true, and it works in Chrome browser on Android 7 and iOS 11. Please clarify, what WCS version do you use, on what mobile platforms and how do you reproduce the issue?
  3. Max

    "rtmp_transponder_stream_name_prefix =rtmp_ " setting

    Hello. The setting rtmp_transponder_stream_name_prefix=rtmp_ is the default, so it is not in the flashphoner.properties file. In your case, to republish WebRTC stream to Facebook and Youtube you should set this parameter to empty in file: rtmp_transponder_stream_name_prefix=
  4. Max

    Issue with SIP audio calls in SIP as RTMP .

    Hello. We reproduced the issue with Plivo.com and raised internal case WCS-1672. We investigate it and let you know when there will be any results.
  5. Max

    Issue with SIP audio calls in SIP as RTMP .

    Hello As we mentioned above we do not schedule time-frames on the current support level. Please send all necessary instructions and accesses which allow us to test this any time. I.e. you can setup a test plivo account and provide access and instructions for this stage account.
  6. Max

    Issue with SIP audio calls in SIP as RTMP .

    Hello If you are testing with plivo, you can create a simple IVR menu, not a conference bridge. Then we will be able to test DTMF with the menu.
  7. Max

    Issue with SIP audio calls in SIP as RTMP .

    Hello We have checked your server according received SSH access. You have version WCS 5.0.x On the current support level (Simple License, Forum support) we do not provide any scheduled support sessions based on a time window. Please 1. Install latest 5.1.x version...
  8. Max

    Issue with SIP audio calls in SIP as RTMP .

    Hello Please provide: 1. tcpdump log captured during your own test: tcpdump udp -s 4096 -w log.pcap 2. SIP call details 1) login 2) authentication name 3) password 4) outbound proxy 5) port 6) callee number (what number we should call to hear reply) 7) DTMF number (what DTMF number we should...
  9. Max

    Whitelabel Error Page

    Hello You have to fix it according our recommendations. We don't manage the server. We just assist and provide recommendations. For now we offer to fix the /etc/hosts file.
  10. Max

    Whitelabel Error Page

    You didn't set hostname to /etc/hosts As you can see, file /etc/hosts does not contain your hostname
  11. Max

    Whitelabel Error Page

    Hello Server core has not been started. 1. Know hostname of your server #hostname 2. Add hostname of your server to /etc/hosts Example of /etc/hosts 127.0.0.1 myhost Here myhost is your hostname on step (1) 3. Start server service webcallserver start
  12. Max

    Issue with SIP audio calls in SIP as RTMP .

    hello try to remove 'telephone-event' from the exclude statement codecs_exclude_sip_rtmp =opus,g729,g722,mpeg4-generic,vp8,mpv,telephone-event It should be codecs_exclude_sip_rtmp =opus,g729,g722,mpeg4-generic,vp8,mpv telephone-event is DTMF
  13. Max

    Web call Server question. It can not run sample on other pc

    Please clarify: 1. You publish a stream on Origin server (cdn_point_of_entry) 2. You cannot play this stream from Origin server or 3. You cannot play this stream from Edge server (some other WCS instance that set up as Edge)?
  14. Max

    Отключать пустые трансляции

    Добрый день. Действительно, Chrome может посылать пустой видеотрафик, если камера занята другим приложением. Периодически эту ошибку исправляют, затем она вновь возникает. Есть два варианта обхода этой ошибки: 1. Видеодорожка, созданная браузером Chrome для занятой камеры, останавливается в...
  15. Max

    CDN 2 Stream Screenshots

    REST query /rest-api/cdn/show_routes was added at least since build 3467. Now, we described it in CDN docs. This query returns current routes list like CLI show cdn-routes command: http://test.flashphoner.com:8081/rest-api/cdn/show_routes { "1-origin2.flashphoner.com-2": "stream1"...
  16. Max

    Web call Server question. It can not run sample on other pc

    Hello, Frank. Yes, you should set the following parameter in WCS_HOME/conf/wcs-manager.properties file -Dmanager.https_port=443 By default, it is set to 8888 and commented with '#' character in the file, so you should uncomment it before you make change.
  17. Max

    Issues with video calls using the Android native mobile app

    Hello, Please try it with VP8 priority to see if that is related to H.264 support. Some devices may not support H.264. In WCS_HOME/conf/flashphoner.properties place vp8 before h264 codecs =opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,vp8,h264,flv,mpv and restart the WCS.
  18. Max

    No webrtc in iOS 12 homescreen app

    Please confirm if the issue is reproducible with the demo server. Tried this on iPhone 5s iOS 12.1, and the issue has not been reproduced: WebRTC not WSPlayer has been used in the case - shortcut to https://demo.flashphoner.com/client2/examples/demo/streaming/player/player.html added to home...
  19. Max

    CDN 2 Stream Screenshots

    Only way to request a snapshot from Edge server is playing a stream from this Edge server. 1. Play 2. Take snapshot You can't just take snapshot from Edge because CDN is implemented lazy way. This means Edge does not have this stream until it is requested by play() request from Web SDK. So...
  20. Max

    No webrtc in iOS 12 homescreen app

    WebRTC has default priority https://demo.flashphoner.com/client2/examples/demo/streaming/player/player.html or https://demo.flashphoner.com/client2/examples/demo/streaming/player/player.html?mediaProvider=WebRTC WSPlayer...
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