Search results

Hello, 1. Please check stream codecs in VLC: Tools - Codec information Supported codecs: MPEG AAC for audio H264 for video 2. Try to switch to none-interleave mode in flashphoner.properties rtsp_interleaved_mode=false This setting requires WCS restart service webcallserver restart

We have tested with latest build 2172 and our Twilio account. 1. It works fine with default settings. 2. It works with sip_as_rtmp_use_new_engine=false Perhaps you have an issue with your Wowza configuration. Please test the build.

Hello, Similar deployment (when ip != ip_local) may require additional setting in flashphoner.properties: rtc_ice_add_local_component=true This setting requires WCS restart. service webcallserver restart Your demo stream works with our demo-player https://wcs5-eu.flashphoner.com/demo2/player

Hello This feature is not currently supported. However we have similar feature in our roadmap.

You can update to previous build 2101 And check if it works for you with: sip_as_rtmp_use_new_engine=false or sip_as_rtmp_use_new_engine=true The issue is under investigation. I will inform you if we have any news.

Добрый день. Для hls на данный момент аутентификации нет. Если порт открыт, к нему можно подключиться по адресу http://host:8082/stream/stream.m3u8 если закрыт, то нельзя.

В соседней теме есть этот же вопрос Это событие инициируется если на канале между WCS сервером и зрителем обнаружено 5% и более потерь. Можно повысить этот порог настройкой (приводится значение по умолчанию) webrtc_cc2_bitrate_overuse_event_threshold=0.05

Добрый день Посмотрите этот код: https://github.com/flashphoner/flashphoner_client/blob/wcs_api-2.0/examples/demo/streaming/player/player.js Чтобы работало в Safari, нужно подключить скрипты WSReceiver2.js и video-worker2.js Flashphoner.init({ flashMediaProviderSwfLocation...

Please read this thread http://forum.flashphoner.com/threads/sip-phone-ui-error.10848/ We have fixed issue with Asterisk callId in the latest build. Perhaps you had similar issue in your integration.

This issue was fixed in the build 2168 Please do service webcallserver update to apply the fix For older builds you may need to perform manual update using 1. Download build 2. ./install.sh

You can try to update to build 2080 and older. Update steps: 1. Stop WCS server. service webcallserver stop 2. Download, unzip and run install script ./install.sh 3. Start WCS server service webcallserver start With these builds you have to change client-side to get this working: Edit file...

We have investigated the issue. As you can see from the image above: Asterisk uses ':' in the callId. This affects WebRTC leg on ICE connection phase. The fix may take some time. We will inform you once it is fixed. You can also try to change Asterisk's callId and remove ':' from callId to get...

We are able to see stun binding errors in the pcap log We can't reproduce these errors on our testing server 2165. Please provide 1. SSH sudo access to your server 2. Two Asterisk SIP accounts for testing. 3. Admin password for WCS dashboard. We will check the same through your server.

WCS sends audio in AAC codecs in latest builds. Perhaps if you deploy the same Wowza version on your stage host, it would work properly. Other options: 1. Switch to old engine in flashphoner.properties: sip_as_rtmp_use_new_engine=false In such case WCS will use G.711 codec received from Twilio...

Это видео работает в iOS Safari rtsp://mobile:mob999@176.114.226.39:5541/cam/realmonitor?channel=1&subtype=1 На iPhone выглядит так: Обратите внимание, что все по http: http://host:9091 ws://host:8080 По https Safari тоже должен работать, но там нужны сертификаты.

The pcap file is incomplete. It does not contain any calls. Please make the pcap file again. 1. Start capture tcpdump udp -s 4096 -w log.pcap 2. Make the call. 3. Stop capture. Press Ctrl + C.

Please prepare following logs 1. Server log /usr/local/FlashphonerWebCallServer/logs/server_logs/flashphoner.log 2. Server dump tcpdump udp -s 4096 -w log.pcap 3. Browser console log. 4. /usr/local/FlashphonerWebCallServer/conf directory Please zip these files and send to logs@flashphoner.com We...

We were able to reproduce this crash with Firefox streamer and IE player. It is fixed in latest build 2165. Update command: service webcallserver update Please report with logs if you still have this crash.

Да, не из-за звука. Профиль H.264 High поддерживается только с транскодированием. Можете включить транскодирование в настройках сервера, как показано выше. Эта камера работает через MPEG-4 Video. Этот формат сейчас не поддерживается: Поддерживаемый формат для видео: H.264 - MPEG-4 AVC:

Нет. Не игнорируются. С именами всех этих настроек мы еще поработаем и опишем в документации окончательный вариант.