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In example above, we enabled G729 codec only. Try to enable more audio codecs. Also, if you do not use video calls at all, you can disable vp8 codec. Also, you can enable TCP usage for SIP signaling with this setting sip_force_tcp=true This eliminates SDP fragmentation, so no codecs exclusion is...

Hello. Unfortunately no, a user action is required to enable sound on autoplay in Chrome. Moreover, Firefox announces the same autoplay policy change in near future. In WCS 5.0 and 5.1, the WCS manager module performed web interface functions and stored admin password in its own database. In...

Сервер demo.flashphoner.com работает.

Please describe how the server is used and send the logs and configuration files to support@flashphoner.com.

WCS update from 5.1 to 5.2 is described here.

Hello. Please update to the latest build from this page and check again. If the issue persists, please prepare a report as described here, including gc-core-* logs and send to support@flashphoner.com, we will check. Also check what top command shows. Anyway, 2.91 Gb for 5 days does not look...

Hello. We fixed authentication issue in build 5.2.39. Please update and check. If problem perststs, please provide access to RTSP camera or prepare report as described here and send to support@flashphoner.com. Make shure you have captured RTSP packets to traffic dump.

Hello. We'll make a fix that should force DTMF sending even if no telephone-event codec in SDP (internal ticket WCS-1860). Also it seems like the problem is in WCS codecs configuration. In flashphoner.properties you have sent default codecs settings are used...

Hello. We work on it (internal ticket WCS-1861) and let you know when we fix {startTime} and add {endTime}

Hello. You can rename next recording part based on last modification time of previuos part. We'll try to reproduce it. Commonly, WCS should not toch a recording file after stream is actually finished. But, for example, pulled RTSP stream finished in 60 seconds after last subscriber is off, and...

Hello. We have tested the latest build 5.2.36 with OpenSIPS and Asterisk, with WCS default settings. WCS sends RFC2833 DTMF correctly, see traffic dump analisys screenshot So please update to this build and provide us traffic dumps for all of your PBXes where WCS does not work: Also, provide...

Your SIP gateway (PBX) answers in SDP SIP/2.0 200 OK m=audio 8376 RTP/AVP 18 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=sendrecv As you can see, your PBX does not indicate telephone-event (DTMF). And therefore your PBX does not work properly. Not WCS. You claim WCS 5.0 sends DTMF properly...

Hello WCS supports two DTMF modes: DTMF over RFC2833 DTMF over SIP INFO Your PBX does not support these modes. Third mode "DTMF Inband" is currently unsupported in WCS server. So to get this working you have to either migrate to another PBX or setup DTMF over RFC2833 / SIP INFO on your PBX...

https://docs.flashphoner.com/display/WCS52EN/Stream+recording#Streamrecording-Formingthenameofthestreamrecordfile You have to pass parameter {startTime}. End time is time of last modification. Therefore if you have a file with starting time stream-300.mp4, you can post-process this file and...

Дайте ссылки для скачивания на несколько mp4 файлов, которые не играют, мы их проверим.

Формирование имени файла записи потока Нужно настроить именование записи с параметром {startTime} Тогда файл записи будет иметь ts начала записи и будет доступен в API.

Проблема в SSL. Сертификат должен быть импортирован либо нужно открывать страницу https://82.165.69.60:8443 чтобы прогрузить сертификат в кэш браузера https://docs.flashphoner.com/display/WCS52RU/Websocket+SSL

Please provide access to RTSP camera stream. We will check. If you can't provide access 1. Make tcpdump tcpdump -i any -B 10000 -w log.pcap 2. Download pcap file, open in Wireshark and make sure you have captured RTSP packets https://docs.flashphoner.com/pages/viewpage.action?pageId=9242002 If...

This issue will be processed according internal priorities. We do not provide any ETA. You can also fix/workaround this issue on PBX end adding "telephone-event" into 200 OK SDP.

Также можете попробовать добавить настройку на стороне сервера: vod_mp4_container_new=true Этот контейнер поддерживает только определенный формат mp4 файлов MOV Проверить формат вашего файла можно как показано здесь.